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You are here: manpages

FFSERVER-ALL

Section: (1)
Updated:
Index Return to Main Contents

NAME

ffserver - ffserver video server

SYNOPSIS

ffserver [options]

DESCRIPTION

ffserver is a streaming server for both audio and video. It supports several live feeds, streaming from files and time shifting on live feeds. You can seek to positions in the past on each live feed, provided you specify a big enough feed storage.

ffserver is configured through a configuration file, which is read at startup. If not explicitly specified, it will read from /etc/ffserver.conf.

ffserver receives prerecorded files or FFM streams from some ffmpeg instance as input, then streams them over RTP/RTSP/HTTP.

An ffserver instance will listen on some port as specified in the configuration file. You can launch one or more instances of ffmpeg and send one or more FFM streams to the port where ffserver is expecting to receive them. Alternately, you can make ffserver launch such ffmpeg instances at startup.

Input streams are called feeds, and each one is specified by a "<Feed>" section in the configuration file.

For each feed you can have different output streams in various formats, each one specified by a "<Stream>" section in the configuration file.

DETAILED DESCRIPTION

ffserver works by forwarding streams encoded by ffmpeg, or pre-recorded streams which are read from disk.

Precisely, ffserver acts as an HTTP server, accepting POST requests from ffmpeg to acquire the stream to publish, and serving RTSP clients or HTTP clients GET requests with the stream media content.

A feed is an FFM stream created by ffmpeg, and sent to a port where ffserver is listening.

Each feed is identified by a unique name, corresponding to the name of the resource published on ffserver, and is configured by a dedicated "Feed" section in the configuration file.

The feed publish URL is given by:

        http://<ffserver_ip_address>:<http_port>/<feed_name>

where ffserver_ip_address is the IP address of the machine where ffserver is installed, http_port is the port number of the HTTP server (configured through the HTTPPort option), and feed_name is the name of the corresponding feed defined in the configuration file.

Each feed is associated to a file which is stored on disk. This stored file is used to send pre-recorded data to a player as fast as possible when new content is added in real-time to the stream.

A ``live-stream'' or ``stream'' is a resource published by ffserver, and made accessible through the HTTP protocol to clients.

A stream can be connected to a feed, or to a file. In the first case, the published stream is forwarded from the corresponding feed generated by a running instance of ffmpeg, in the second case the stream is read from a pre-recorded file.

Each stream is identified by a unique name, corresponding to the name of the resource served by ffserver, and is configured by a dedicated "Stream" section in the configuration file.

The stream access HTTP URL is given by:

        http://<ffserver_ip_address>:<http_port>/<stream_name>[<options>]

The stream access RTSP URL is given by:

        http://<ffserver_ip_address>:<rtsp_port>/<stream_name>[<options>]

stream_name is the name of the corresponding stream defined in the configuration file. options is a list of options specified after the URL which affects how the stream is served by ffserver. http_port and rtsp_port are the HTTP and RTSP ports configured with the options HTTPPort and RTSPPort respectively.

In case the stream is associated to a feed, the encoding parameters must be configured in the stream configuration. They are sent to ffmpeg when setting up the encoding. This allows ffserver to define the encoding parameters used by the ffmpeg encoders.

The ffmpeg override_ffserver commandline option allows one to override the encoding parameters set by the server.

Multiple streams can be connected to the same feed.

For example, you can have a situation described by the following graph:

                       _________       __________
                      |         |     |          |
        ffmpeg 1 -----| feed 1  |-----| stream 1 |
            \         |_________|\    |__________|
             \                    \
              \                    \   __________
               \                    \ |          |
                \                    \| stream 2 |
                 \                    |__________|
                  \
                   \   _________       __________
                    \ |         |     |          |
                     \| feed 2  |-----| stream 3 |
                      |_________|     |__________|
        
                       _________       __________
                      |         |     |          |
        ffmpeg 2 -----| feed 3  |-----| stream 4 |
                      |_________|     |__________|
        
                       _________       __________
                      |         |     |          |
                      | file 1  |-----| stream 5 |
                      |_________|     |__________|

FFM, FFM2 formats

FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of video and audio streams and encoding options, and can store a moving time segment of an infinite movie or a whole movie.

FFM is version specific, and there is limited compatibility of FFM files generated by one version of ffmpeg/ffserver and another version of ffmpeg/ffserver. It may work but it is not guaranteed to work.

FFM2 is extensible while maintaining compatibility and should work between differing versions of tools. FFM2 is the default.

Status stream

ffserver supports an HTTP interface which exposes the current status of the server.

Simply point your browser to the address of the special status stream specified in the configuration file.

For example if you have:

        <Stream status.html>
        Format status
        
        # Only allow local people to get the status
        ACL allow localhost
        ACL allow 192.168.0.0 192.168.255.255
        </Stream>

then the server will post a page with the status information when the special stream status.html is requested.

How do I make it work?

As a simple test, just run the following two command lines where INPUTFILE is some file which you can decode with ffmpeg:

        ffserver -f doc/ffserver.conf &
        ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm

At this point you should be able to go to your Windows machine and fire up Windows Media Player (WMP). Go to Open URL and enter

            http://<linuxbox>:8090/test.asf

You should (after a short delay) see video and hear audio.

WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to transfer the entire file before starting to play. The same is true of AVI files.

You should edit the ffserver.conf file to suit your needs (in terms of frame rates etc). Then install ffserver and ffmpeg, write a script to start them up, and off you go.

What else can it do?

You can replay video from .ffm files that was recorded earlier. However, there are a number of caveats, including the fact that the ffserver parameters must match the original parameters used to record the file. If they do not, then ffserver deletes the file before recording into it. (Now that I write this, it seems broken).

You can fiddle with many of the codec choices and encoding parameters, and there are a bunch more parameters that you cannot control. Post a message to the mailing list if there are some 'must have' parameters. Look in ffserver.conf for a list of the currently available controls.

It will automatically generate the ASX or RAM files that are often used in browsers. These files are actually redirections to the underlying ASF or RM file. The reason for this is that the browser often fetches the entire file before starting up the external viewer. The redirection files are very small and can be transferred quickly. [The stream itself is often 'infinite' and thus the browser tries to download it and never finishes.]

Tips

* When you connect to a live stream, most players (WMP, RA, etc) want to buffer a certain number of seconds of material so that they can display the signal continuously. However, ffserver (by default) starts sending data in realtime. This means that there is a pause of a few seconds while the buffering is being done by the player. The good news is that this can be cured by adding a '?buffer=5' to the end of the URL. This means that the stream should start 5 seconds in the past --- and so the first 5 seconds of the stream are sent as fast as the network will allow. It will then slow down to real time. This noticeably improves the startup experience.

You can also add a 'Preroll 15' statement into the ffserver.conf that will add the 15 second prebuffering on all requests that do not otherwise specify a time. In addition, ffserver will skip frames until a key_frame is found. This further reduces the startup delay by not transferring data that will be discarded.

Why does the ?buffer / Preroll stop working after a time?

It turns out that (on my machine at least) the number of frames successfully grabbed is marginally less than the number that ought to be grabbed. This means that the timestamp in the encoded data stream gets behind realtime. This means that if you say 'Preroll 10', then when the stream gets 10 or more seconds behind, there is no Preroll left.

Fixing this requires a change in the internals of how timestamps are handled.

Does the ?date= stuff work.

Yes (subject to the limitation outlined above). Also note that whenever you start ffserver, it deletes the ffm file (if any parameters have changed), thus wiping out what you had recorded before.

The format of the "?date=xxxxxx" is fairly flexible. You should use one of the following formats (the 'T' is literal):

        * YYYY-MM-DDTHH:MM:SS     (localtime)
        * YYYY-MM-DDTHH:MM:SSZ    (UTC)

You can omit the YYYY-MM-DD, and then it refers to the current day. However note that ?date=16:00:00 refers to 16:00 on the current day --- this may be in the future and so is unlikely to be useful.

You use this by adding the ?date= to the end of the URL for the stream. For example: http://localhost:8080/test.asf?date=2002-07-26T23:05:00.

OPTIONS

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

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

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

Stream specifiers

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

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

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

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

Possible forms of stream specifiers are:

stream_index: Matches the stream with this index. E.g. "-threads:1 4" would set the thread count for the second stream to 4.
stream_type[:stream_index]: stream_type is one of following: 'v' or 'V' for video, 'a' for audio, 's' for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video streams, 'V' only matches video streams which are not attached pictures, video thumbnails or cover arts. If stream_index is given, then it matches stream number stream_index of this type. Otherwise, it matches all streams of this type.
p:program_id[:stream_index]: If stream_index is given, then it matches the stream with number stream_index in the program with the id program_id. Otherwise, it matches all streams in the program.
#stream_id or i:stream_id: Match the stream by stream id (e.g. PID in MPEG-TS container).
m:key[:value]: Matches streams with the metadata tag key having the specified value. If value is not given, matches streams that contain the given tag with any value.
u: Matches streams with usable configuration, the codec must be defined and the essential information such as video dimension or audio sample rate must be present.
Note that in ffmpeg, matching by metadata will only work properly for input files.

Generic options

These options are shared amongst the ff* tools.

-L

Show license.

-h, -?, -help, --help [arg]

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

Possible values of arg are:

long: Print advanced tool options in addition to the basic tool options.
full: Print complete list of options, including shared and private options for encoders, decoders, demuxers, muxers, filters, etc.
decoder=decoder_name: Print detailed information about the decoder named decoder_name. Use the -decoders option to get a list of all decoders.
encoder=encoder_name: Print detailed information about the encoder named encoder_name. Use the -encoders option to get a list of all encoders.
demuxer=demuxer_name: Print detailed information about the demuxer named demuxer_name. Use the -formats option to get a list of all demuxers and muxers.
muxer=muxer_name: Print detailed information about the muxer named muxer_name. Use the -formats option to get a list of all muxers and demuxers.
filter=filter_name: Print detailed information about the filter name filter_name. Use the -filters option to get a list of all filters.

-version

Show version.

-formats

Show available formats (including devices).

-devices

Show available devices.

-codecs

Show all codecs known to libavcodec.

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

-decoders

Show available decoders.

-encoders

Show all available encoders.

-bsfs

Show available bitstream filters.

-protocols

Show available protocols.

-filters

Show available libavfilter filters.

-pix_fmts

Show available pixel formats.

-sample_fmts

Show available sample formats.

-layouts

Show channel names and standard channel layouts.

-colors

Show recognized color names.

-sources device[,opt1=val1[,opt2=val2]...]

Show autodetected sources of the intput device. Some devices may provide system-dependent source names that cannot be autodetected. The returned list cannot be assumed to be always complete.

        ffmpeg -sources pulse,server=192.168.0.4

-sinks device[,opt1=val1[,opt2=val2]...]

Show autodetected sinks of the output device. Some devices may provide system-dependent sink names that cannot be autodetected. The returned list cannot be assumed to be always complete.

        ffmpeg -sinks pulse,server=192.168.0.4

-loglevel [repeat+]loglevel | -v [repeat+]loglevel

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

quiet, -8: Show nothing at all; be silent.
panic, 0: Only show fatal errors which could lead the process to crash, such as an assertion failure. This is not currently used for anything.
fatal, 8: Only show fatal errors. These are errors after which the process absolutely cannot continue.
error, 16: Show all errors, including ones which can be recovered from.
warning, 24: Show all warnings and errors. Any message related to possibly incorrect or unexpected events will be shown.
info, 32: Show informative messages during processing. This is in addition to warnings and errors. This is the default value.
verbose, 40: Same as "info", except more verbose.
debug, 48: Show everything, including debugging information.
trace, 56

: By default the program logs to stderr. If coloring is supported by the terminal, colors are used to mark errors and warnings. Log coloring can be disabled setting the environment variable AV_LOG_FORCE_NOCOLOR or NO_COLOR, or can be forced setting the environment variable AV_LOG_FORCE_COLOR. The use of the environment variable NO_COLOR is deprecated and will be dropped in a future FFmpeg version.

-report

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

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

The following options are recognized:

file: set the file name to use for the report; %p is expanded to the name of the program, %t is expanded to a timestamp, "%%" is expanded to a plain "%"
level: set the log verbosity level using a numerical value (see "-loglevel").

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

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

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

-hide_banner

Suppress printing banner.

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

-cpuflags flags (global)

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

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

Possible flags for this option are:

x86

mmx
mmxext
sse
sse2
sse2slow
sse3
sse3slow
ssse3
atom
sse4.1
sse4.2
avx
avx2
xop
fma3
fma4
3dnow
3dnowext
bmi1
bmi2
cmov

ARM

armv5te
armv6
armv6t2
vfp
vfpv3
neon
setend

AArch64

armv8
vfp
neon

PowerPC

altivec

Specific Processors

pentium2
pentium3
pentium4
k6
k62
athlon
athlonxp
k8

-opencl_bench

This option is used to benchmark all available OpenCL devices and print the results. This option is only available when FFmpeg has been compiled with "--enable-opencl".

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

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

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

Run the command:

        ffmpeg -opencl_bench

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

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

-opencl_options options (global)

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

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

AVOptions

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

generic: These options can be set for any container, codec or device. Generic options are listed under AVFormatContext options for containers/devices and under AVCodecContext options for codecs.
private: These options are specific to the given container, device or codec. Private options are listed under their corresponding containers/devices/codecs.

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

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

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

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

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

Main options

-f configfile: Read configuration file configfile. If not specified it will read by default from /etc/ffserver.conf.
-n: Enable no-launch mode. This option disables all the "Launch" directives within the various "<Feed>" sections. Since ffserver will not launch any ffmpeg instances, you will have to launch them manually.
-d: Enable debug mode. This option increases log verbosity, and directs log messages to stdout. When specified, the CustomLog option is ignored.

CONFIGURATION FILE SYNTAX

ffserver reads a configuration file containing global options and settings for each stream and feed.

The configuration file consists of global options and dedicated sections, which must be introduced by "<SECTION_NAME ARGS>`` on a separate line and must be terminated by a line in the form ''</SECTION_NAME>". ARGS is optional.

Currently the following sections are recognized: Feed, Stream, Redirect.

A line starting with "#" is ignored and treated as a comment.

Name of options and sections are case-insensitive.

ACL syntax

An ACL (Access Control List) specifies the address which are allowed to access a given stream, or to write a given feed.

It accepts the folling forms

*

Allow/deny access to address.

        ACL ALLOW <address>
        ACL DENY <address>

*

Allow/deny access to ranges of addresses from first_address to last_address.

        ACL ALLOW <first_address> <last_address>
        ACL DENY <first_address> <last_address>

You can repeat the ACL allow/deny as often as you like. It is on a per stream basis. The first match defines the action. If there are no matches, then the default is the inverse of the last ACL statement.

Thus 'ACL allow localhost' only allows access from localhost. 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and allow everybody else.

Global options

HTTPPort port_number

Port port_number

RTSPPort port_number

HTTPPort sets the HTTP server listening TCP port number, RTSPPort sets the RTSP server listening TCP port number.

Port is the equivalent of HTTPPort and is deprecated.

You must select a different port from your standard HTTP web server if it is running on the same computer.

If not specified, no corresponding server will be created.

HTTPBindAddress ip_address

BindAddress ip_address

RTSPBindAddress ip_address

Set address on which the HTTP/RTSP server is bound. Only useful if you have several network interfaces.

BindAddress is the equivalent of HTTPBindAddress and is deprecated.

MaxHTTPConnections n

Set number of simultaneous HTTP connections that can be handled. It has to be defined before the MaxClients parameter, since it defines the MaxClients maximum limit.

Default value is 2000.

MaxClients n

Set number of simultaneous requests that can be handled. Since ffserver is very fast, it is more likely that you will want to leave this high and use MaxBandwidth.

Default value is 5.

MaxBandwidth kbps

Set the maximum amount of kbit/sec that you are prepared to consume when streaming to clients.

Default value is 1000.

CustomLog filename

Set access log file (uses standard Apache log file format). '-' is the standard output.

If not specified ffserver will produce no log.

In case the commandline option -d is specified this option is ignored, and the log is written to standard output.

NoDaemon

Set no-daemon mode. This option is currently ignored since now ffserver will always work in no-daemon mode, and is deprecated.

UseDefaults

NoDefaults

Control whether default codec options are used for the all streams or not. Each stream may overwrite this setting for its own. Default is UseDefaults. The lastest occurrence overrides previous if multiple definitions.

Feed section

A Feed section defines a feed provided to ffserver.

Each live feed contains one video and/or audio sequence coming from an ffmpeg encoder or another ffserver. This sequence may be encoded simultaneously with several codecs at several resolutions.

A feed instance specification is introduced by a line in the form:

        <Feed FEED_FILENAME>

where FEED_FILENAME specifies the unique name of the FFM stream.

The following options are recognized within a Feed section.

File filename

ReadOnlyFile filename

Set the path where the feed file is stored on disk.

If not specified, the /tmp/FEED.ffm is assumed, where FEED is the feed name.

If ReadOnlyFile is used the file is marked as read-only and it will not be deleted or updated.

Truncate

Truncate the feed file, rather than appending to it. By default ffserver will append data to the file, until the maximum file size value is reached (see FileMaxSize option).

FileMaxSize size

Set maximum size of the feed file in bytes. 0 means unlimited. The postfixes "K" (2^10), "M" (2^20), and "G" (2^30) are recognized.

Default value is 5M.

Launch args

Launch an ffmpeg command when creating ffserver.

args must be a sequence of arguments to be provided to an ffmpeg instance. The first provided argument is ignored, and it is replaced by a path with the same dirname of the ffserver instance, followed by the remaining argument and terminated with a path corresponding to the feed.

When the launched process exits, ffserver will launch another program instance.

In case you need a more complex ffmpeg configuration, e.g. if you need to generate multiple FFM feeds with a single ffmpeg instance, you should launch ffmpeg by hand.

This option is ignored in case the commandline option -n is specified.

ACL spec

Specify the list of IP address which are allowed or denied to write the feed. Multiple ACL options can be specified.

Stream section

A Stream section defines a stream provided by ffserver, and identified by a single name.

The stream is sent when answering a request containing the stream name.

A stream section must be introduced by the line:

        <Stream STREAM_NAME>

where STREAM_NAME specifies the unique name of the stream.

The following options are recognized within a Stream section.

Encoding options are marked with the encoding tag, and they are used to set the encoding parameters, and are mapped to libavcodec encoding options. Not all encoding options are supported, in particular it is not possible to set encoder private options. In order to override the encoding options specified by ffserver, you can use the ffmpeg override_ffserver commandline option.

Only one of the Feed and File options should be set.

Feed feed_name: Set the input feed. feed_name must correspond to an existing feed defined in a "Feed" section.
When this option is set, encoding options are used to setup the encoding operated by the remote ffmpeg process.
File filename: Set the filename of the pre-recorded input file to stream.
When this option is set, encoding options are ignored and the input file content is re-streamed as is.
Format format_name: Set the format of the output stream.
Must be the name of a format recognized by FFmpeg. If set to status, it is treated as a status stream.
InputFormat format_name: Set input format. If not specified, it is automatically guessed.
Preroll n: Set this to the number of seconds backwards in time to start. Note that most players will buffer 5-10 seconds of video, and also you need to allow for a keyframe to appear in the data stream.
Default value is 0.
StartSendOnKey: Do not send stream until it gets the first key frame. By default ffserver will send data immediately.
MaxTime n: Set the number of seconds to run. This value set the maximum duration of the stream a client will be able to receive.
A value of 0 means that no limit is set on the stream duration.
ACL spec: Set ACL for the stream.
DynamicACL spec
RTSPOption option
MulticastAddress address
MulticastPort port
MulticastTTL integer
NoLoop
FaviconURL url: Set favicon (favourite icon) for the server status page. It is ignored for regular streams.
Author value
Comment value
Copyright value
Title value: Set metadata corresponding to the option. All these options are deprecated in favor of Metadata.
Metadata key value: Set metadata value on the output stream.
UseDefaults
NoDefaults: Control whether default codec options are used for the stream or not. Default is UseDefaults unless disabled globally.
NoAudio
NoVideo: Suppress audio/video.
AudioCodec codec_name (encoding,audio): Set audio codec.
AudioBitRate rate (encoding,audio): Set bitrate for the audio stream in kbits per second.
AudioChannels n (encoding,audio): Set number of audio channels.
AudioSampleRate n (encoding,audio): Set sampling frequency for audio. When using low bitrates, you should lower this frequency to 22050 or 11025. The supported frequencies depend on the selected audio codec.
AVOptionAudio [codec:]option value (encoding,audio): Set generic or private option for audio stream. Private option must be prefixed with codec name or codec must be defined before.
AVPresetAudio preset (encoding,audio): Set preset for audio stream.
VideoCodec codec_name (encoding,video): Set video codec.
VideoBitRate n (encoding,video): Set bitrate for the video stream in kbits per second.
VideoBitRateRange range (encoding,video): Set video bitrate range.
A range must be specified in the form minrate-maxrate, and specifies the minrate and maxrate encoding options expressed in kbits per second.
VideoBitRateRangeTolerance n (encoding,video): Set video bitrate tolerance in kbits per second.
PixelFormat pixel_format (encoding,video): Set video pixel format.
Debug integer (encoding,video): Set video debug encoding option.
Strict integer (encoding,video): Set video strict encoding option.
VideoBufferSize n (encoding,video): Set ratecontrol buffer size, expressed in KB.
VideoFrameRate n (encoding,video): Set number of video frames per second.
VideoSize (encoding,video): Set size of the video frame, must be an abbreviation or in the form WxH. See the Video size section in the ffmpeg-utils(1) manual.
Default value is "160x128".
VideoIntraOnly (encoding,video): Transmit only intra frames (useful for low bitrates, but kills frame rate).
VideoGopSize n (encoding,video): If non-intra only, an intra frame is transmitted every VideoGopSize frames. Video synchronization can only begin at an intra frame.
VideoTag tag (encoding,video): Set video tag.
VideoHighQuality (encoding,video)
Video4MotionVector (encoding,video)
BitExact (encoding,video): Set bitexact encoding flag.
IdctSimple (encoding,video): Set simple IDCT algorithm.
Qscale n (encoding,video): Enable constant quality encoding, and set video qscale (quantization scale) value, expressed in n QP units.
VideoQMin n (encoding,video)
VideoQMax n (encoding,video): Set video qmin/qmax.
VideoQDiff integer (encoding,video): Set video qdiff encoding option.
LumiMask float (encoding,video)
DarkMask float (encoding,video): Set lumi_mask/dark_mask encoding options.
AVOptionVideo [codec:]option value (encoding,video): Set generic or private option for video stream. Private option must be prefixed with codec name or codec must be defined before.
AVPresetVideo preset (encoding,video): Set preset for video stream.
preset must be the path of a preset file.

Server status stream

A server status stream is a special stream which is used to show statistics about the ffserver operations.

It must be specified setting the option Format to status.

Redirect section

A redirect section specifies where to redirect the requested URL to another page.

A redirect section must be introduced by the line:

        <Redirect NAME>

where NAME is the name of the page which should be redirected.

It only accepts the option URL, which specify the redirection URL.

STREAM EXAMPLES

*

Multipart JPEG

        <Stream test.mjpg>
        Feed feed1.ffm
        Format mpjpeg
        VideoFrameRate 2
        VideoIntraOnly
        NoAudio
        Strict -1
        </Stream>

*

Single JPEG

        <Stream test.jpg>
        Feed feed1.ffm
        Format jpeg
        VideoFrameRate 2
        VideoIntraOnly
        VideoSize 352x240
        NoAudio
        Strict -1
        </Stream>

*

Flash

        <Stream test.swf>
        Feed feed1.ffm
        Format swf
        VideoFrameRate 2
        VideoIntraOnly
        NoAudio
        </Stream>

*

ASF compatible

        <Stream test.asf>
        Feed feed1.ffm
        Format asf
        VideoFrameRate 15
        VideoSize 352x240
        VideoBitRate 256
        VideoBufferSize 40
        VideoGopSize 30
        AudioBitRate 64
        StartSendOnKey
        </Stream>

*

MP3 audio

        <Stream test.mp3>
        Feed feed1.ffm
        Format mp2
        AudioCodec mp3
        AudioBitRate 64
        AudioChannels 1
        AudioSampleRate 44100
        NoVideo
        </Stream>

*

Ogg Vorbis audio

        <Stream test.ogg>
        Feed feed1.ffm
        Metadata title "Stream title"
        AudioBitRate 64
        AudioChannels 2
        AudioSampleRate 44100
        NoVideo
        </Stream>

*

Real with audio only at 32 kbits

        <Stream test.ra>
        Feed feed1.ffm
        Format rm
        AudioBitRate 32
        NoVideo
        </Stream>

*

Real with audio and video at 64 kbits

        <Stream test.rm>
        Feed feed1.ffm
        Format rm
        AudioBitRate 32
        VideoBitRate 128
        VideoFrameRate 25
        VideoGopSize 25
        </Stream>

*

For stream coming from a file: you only need to set the input filename and optionally a new format.

        <Stream file.rm>
        File "/usr/local/httpd/htdocs/tlive.rm"
        NoAudio
        </Stream>


        
        <Stream file.asf>
        File "/usr/local/httpd/htdocs/test.asf"
        NoAudio
        Metadata author "Me"
        Metadata copyright "Super MegaCorp"
        Metadata title "Test stream from disk"
        Metadata comment "Test comment"
        </Stream>

SYNTAX

This section documents the syntax and formats employed by the FFmpeg libraries and tools.

Quoting and escaping

FFmpeg adopts the following quoting and escaping mechanism, unless explicitly specified. The following rules are applied:

*: ' and \ are special characters (respectively used for quoting and escaping). In addition to them, there might be other special characters depending on the specific syntax where the escaping and quoting are employed.
*: A special character is escaped by prefixing it with a \.
*: All characters enclosed between '' are included literally in the parsed string. The quote character ' itself cannot be quoted, so you may need to close the quote and escape it.
*: Leading and trailing whitespaces, unless escaped or quoted, are removed from the parsed string.

Note that you may need to add a second level of escaping when using the command line or a script, which depends on the syntax of the adopted shell language.

The function "av_get_token" defined in libavutil/avstring.h can be used to parse a token quoted or escaped according to the rules defined above.

The tool tools/ffescape in the FFmpeg source tree can be used to automatically quote or escape a string in a script.

Examples

*

Escape the string "Crime d'Amour" containing the "'" special character:

        Crime d\'Amour

*

The string above contains a quote, so the "'" needs to be escaped when quoting it:

        'Crime d'\''Amour'

*

Include leading or trailing whitespaces using quoting:

        '  this string starts and ends with whitespaces  '

*

Escaping and quoting can be mixed together:

        ' The string '\'string\'' is a string '

*

To include a literal \ you can use either escaping or quoting:

        'c:\foo' can be written as c:\\foo

Date

The accepted syntax is:

        [(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
        now

If the value is ``now'' it takes the current time.

Time is local time unless Z is appended, in which case it is interpreted as UTC. If the year-month-day part is not specified it takes the current year-month-day.

Time duration

There are two accepted syntaxes for expressing time duration.

        [-][<HH>:]<MM>:<SS>[.<m>...]

HH expresses the number of hours, MM the number of minutes for a maximum of 2 digits, and SS the number of seconds for a maximum of 2 digits. The m at the end expresses decimal value for SS.

        [-]<S>+[.<m>...]

S expresses the number of seconds, with the optional decimal part m.

In both expressions, the optional - indicates negative duration.

Examples

The following examples are all valid time duration:

55: 55 seconds
12:03:45: 12 hours, 03 minutes and 45 seconds
23.189: 23.189 seconds

Video size

Specify the size of the sourced video, it may be a string of the form widthxheight, or the name of a size abbreviation.

The following abbreviations are recognized:

ntsc: 720x480
pal: 720x576
qntsc: 352x240
qpal: 352x288
sntsc: 640x480
spal: 768x576
film: 352x240
ntsc-film: 352x240
sqcif: 128x96
qcif: 176x144
cif: 352x288
4cif: 704x576
16cif: 1408x1152
qqvga: 160x120
qvga: 320x240
vga: 640x480
svga: 800x600
xga: 1024x768
uxga: 1600x1200
qxga: 2048x1536
sxga: 1280x1024
qsxga: 2560x2048
hsxga: 5120x4096
wvga: 852x480
wxga: 1366x768
wsxga: 1600x1024
wuxga: 1920x1200
woxga: 2560x1600
wqsxga: 3200x2048
wquxga: 3840x2400
whsxga: 6400x4096
whuxga: 7680x4800
cga: 320x200
ega: 640x350
hd480: 852x480
hd720: 1280x720
hd1080: 1920x1080
2k: 2048x1080
2kflat: 1998x1080
2kscope: 2048x858
4k: 4096x2160
4kflat: 3996x2160
4kscope: 4096x1716
nhd: 640x360
hqvga: 240x160
wqvga: 400x240
fwqvga: 432x240
hvga: 480x320
qhd: 960x540
2kdci: 2048x1080
4kdci: 4096x2160
uhd2160: 3840x2160
uhd4320: 7680x4320

Video rate

Specify the frame rate of a video, expressed as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation.

The following abbreviations are recognized:

ntsc: 30000/1001
pal: 25/1
qntsc: 30000/1001
qpal: 25/1
sntsc: 30000/1001
spal: 25/1
film: 24/1
ntsc-film: 24000/1001

Ratio

A ratio can be expressed as an expression, or in the form numerator:denominator.

Note that a ratio with infinite (1/0) or negative value is considered valid, so you should check on the returned value if you want to exclude those values.

The undefined value can be expressed using the ``0:0'' string.

Color

It can be the name of a color as defined below (case insensitive match) or a "[0x|#]RRGGBB[AA]" sequence, possibly followed by @ and a string representing the alpha component.

The alpha component may be a string composed by ``0x'' followed by an hexadecimal number or a decimal number between 0.0 and 1.0, which represents the opacity value (0x00 or 0.0 means completely transparent, 0xff or 1.0 completely opaque). If the alpha component is not specified then 0xff is assumed.

The string random will result in a random color.

The following names of colors are recognized:

AliceBlue: 0xF0F8FF
AntiqueWhite: 0xFAEBD7
Aqua: 0x00FFFF
Aquamarine: 0x7FFFD4
Azure: 0xF0FFFF
Beige: 0xF5F5DC
Bisque: 0xFFE4C4
Black: 0x000000
BlanchedAlmond: 0xFFEBCD
Blue: 0x0000FF
BlueViolet: 0x8A2BE2
Brown: 0xA52A2A
BurlyWood: 0xDEB887
CadetBlue: 0x5F9EA0
Chartreuse: 0x7FFF00
Chocolate: 0xD2691E
Coral: 0xFF7F50
CornflowerBlue: 0x6495ED
Cornsilk: 0xFFF8DC
Crimson: 0xDC143C
Cyan: 0x00FFFF
DarkBlue: 0x00008B
DarkCyan: 0x008B8B
DarkGoldenRod: 0xB8860B
DarkGray: 0xA9A9A9
DarkGreen: 0x006400
DarkKhaki: 0xBDB76B
DarkMagenta: 0x8B008B
DarkOliveGreen: 0x556B2F
Darkorange: 0xFF8C00
DarkOrchid: 0x9932CC
DarkRed: 0x8B0000
DarkSalmon: 0xE9967A
DarkSeaGreen: 0x8FBC8F
DarkSlateBlue: 0x483D8B
DarkSlateGray: 0x2F4F4F
DarkTurquoise: 0x00CED1
DarkViolet: 0x9400D3
DeepPink: 0xFF1493
DeepSkyBlue: 0x00BFFF
DimGray: 0x696969
DodgerBlue: 0x1E90FF
FireBrick: 0xB22222
FloralWhite: 0xFFFAF0
ForestGreen: 0x228B22
Fuchsia: 0xFF00FF
Gainsboro: 0xDCDCDC
GhostWhite: 0xF8F8FF
Gold: 0xFFD700
GoldenRod: 0xDAA520
Gray: 0x808080
Green: 0x008000
GreenYellow: 0xADFF2F
HoneyDew: 0xF0FFF0
HotPink: 0xFF69B4
IndianRed: 0xCD5C5C
Indigo: 0x4B0082
Ivory: 0xFFFFF0
Khaki: 0xF0E68C
Lavender: 0xE6E6FA
LavenderBlush: 0xFFF0F5
LawnGreen: 0x7CFC00
LemonChiffon: 0xFFFACD
LightBlue: 0xADD8E6
LightCoral: 0xF08080
LightCyan: 0xE0FFFF
LightGoldenRodYellow: 0xFAFAD2
LightGreen: 0x90EE90
LightGrey: 0xD3D3D3
LightPink: 0xFFB6C1
LightSalmon: 0xFFA07A
LightSeaGreen: 0x20B2AA
LightSkyBlue: 0x87CEFA
LightSlateGray: 0x778899
LightSteelBlue: 0xB0C4DE
LightYellow: 0xFFFFE0
Lime: 0x00FF00
LimeGreen: 0x32CD32
Linen: 0xFAF0E6
Magenta: 0xFF00FF
Maroon: 0x800000
MediumAquaMarine: 0x66CDAA
MediumBlue: 0x0000CD
MediumOrchid: 0xBA55D3
MediumPurple: 0x9370D8
MediumSeaGreen: 0x3CB371
MediumSlateBlue: 0x7B68EE
MediumSpringGreen: 0x00FA9A
MediumTurquoise: 0x48D1CC
MediumVioletRed: 0xC71585
MidnightBlue: 0x191970
MintCream: 0xF5FFFA
MistyRose: 0xFFE4E1
Moccasin: 0xFFE4B5
NavajoWhite: 0xFFDEAD
Navy: 0x000080
OldLace: 0xFDF5E6
Olive: 0x808000
OliveDrab: 0x6B8E23
Orange: 0xFFA500
OrangeRed: 0xFF4500
Orchid: 0xDA70D6
PaleGoldenRod: 0xEEE8AA
PaleGreen: 0x98FB98
PaleTurquoise: 0xAFEEEE
PaleVioletRed: 0xD87093
PapayaWhip: 0xFFEFD5
PeachPuff: 0xFFDAB9
Peru: 0xCD853F
Pink: 0xFFC0CB
Plum: 0xDDA0DD
PowderBlue: 0xB0E0E6
Purple: 0x800080
Red: 0xFF0000
RosyBrown: 0xBC8F8F
RoyalBlue: 0x4169E1
SaddleBrown: 0x8B4513
Salmon: 0xFA8072
SandyBrown: 0xF4A460
SeaGreen: 0x2E8B57
SeaShell: 0xFFF5EE
Sienna: 0xA0522D
Silver: 0xC0C0C0
SkyBlue: 0x87CEEB
SlateBlue: 0x6A5ACD
SlateGray: 0x708090
Snow: 0xFFFAFA
SpringGreen: 0x00FF7F
SteelBlue: 0x4682B4
Tan: 0xD2B48C
Teal: 0x008080
Thistle: 0xD8BFD8
Tomato: 0xFF6347
Turquoise: 0x40E0D0
Violet: 0xEE82EE
Wheat: 0xF5DEB3
White: 0xFFFFFF
WhiteSmoke: 0xF5F5F5
Yellow: 0xFFFF00
YellowGreen: 0x9ACD32

Channel Layout

A channel layout specifies the spatial disposition of the channels in a multi-channel audio stream. To specify a channel layout, FFmpeg makes use of a special syntax.

Individual channels are identified by an id, as given by the table below:

FL: front left
FR: front right
FC: front center
LFE: low frequency
BL: back left
BR: back right
FLC: front left-of-center
FRC: front right-of-center
BC: back center
SL: side left
SR: side right
TC: top center
TFL: top front left
TFC: top front center
TFR: top front right
TBL: top back left
TBC: top back center
TBR: top back right
DL: downmix left
DR: downmix right
WL: wide left
WR: wide right
SDL: surround direct left
SDR: surround direct right
LFE2: low frequency 2

Standard channel layout compositions can be specified by using the following identifiers:

mono: FC
stereo: FL+FR
2.1: FL+FR+LFE
3.0: FL+FR+FC
3.0(back): FL+FR+BC
4.0: FL+FR+FC+BC
quad: FL+FR+BL+BR
quad(side): FL+FR+SL+SR
3.1: FL+FR+FC+LFE
5.0: FL+FR+FC+BL+BR
5.0(side): FL+FR+FC+SL+SR
4.1: FL+FR+FC+LFE+BC
5.1: FL+FR+FC+LFE+BL+BR
5.1(side): FL+FR+FC+LFE+SL+SR
6.0: FL+FR+FC+BC+SL+SR
6.0(front): FL+FR+FLC+FRC+SL+SR
hexagonal: FL+FR+FC+BL+BR+BC
6.1: FL+FR+FC+LFE+BC+SL+SR
6.1: FL+FR+FC+LFE+BL+BR+BC
6.1(front): FL+FR+LFE+FLC+FRC+SL+SR
7.0: FL+FR+FC+BL+BR+SL+SR
7.0(front): FL+FR+FC+FLC+FRC+SL+SR
7.1: FL+FR+FC+LFE+BL+BR+SL+SR
7.1(wide): FL+FR+FC+LFE+BL+BR+FLC+FRC
7.1(wide-side): FL+FR+FC+LFE+FLC+FRC+SL+SR
octagonal: FL+FR+FC+BL+BR+BC+SL+SR
downmix: DL+DR

A custom channel layout can be specified as a sequence of terms, separated by '+' or '|'. Each term can be:

*: the name of a standard channel layout (e.g. mono, stereo, 4.0, quad, 5.0, etc.)
*: the name of a single channel (e.g. FL, FR, FC, LFE, etc.)
*: a number of channels, in decimal, optionally followed by 'c', yielding the default channel layout for that number of channels (see the function "av_get_default_channel_layout")
*: a channel layout mask, in hexadecimal starting with ``0x'' (see the "AV_CH_*" macros in libavutil/channel_layout.h.

Starting from libavutil version 53 the trailing character ``c'' to specify a number of channels will be required, while a channel layout mask could also be specified as a decimal number (if and only if not followed by ``c'').

See also the function "av_get_channel_layout" defined in libavutil/channel_layout.h.

EXPRESSION EVALUATION

When evaluating an arithmetic expression, FFmpeg uses an internal formula evaluator, implemented through the libavutil/eval.h interface.

An expression may contain unary, binary operators, constants, and functions.

Two expressions expr1 and expr2 can be combined to form another expression "expr1;expr2". expr1 and expr2 are evaluated in turn, and the new expression evaluates to the value of expr2.

The following binary operators are available: "+", "-", "*", "/", "^".

The following unary operators are available: "+", "-".

The following functions are available:

abs(x)

Compute absolute value of x.

acos(x)

Compute arccosine of x.

asin(x)

Compute arcsine of x.

atan(x)

Compute arctangent of x.

between(x, min, max)

Return 1 if x is greater than or equal to min and lesser than or equal to max, 0 otherwise.

bitand(x, y)

bitor(x, y)

Compute bitwise and/or operation on x and y.

The results of the evaluation of x and y are converted to integers before executing the bitwise operation.

Note that both the conversion to integer and the conversion back to floating point can lose precision. Beware of unexpected results for large numbers (usually 2^53 and larger).

ceil(expr)

Round the value of expression expr upwards to the nearest integer. For example, ``ceil(1.5)'' is ``2.0''.

clip(x, min, max)

Return the value of x clipped between min and max.

cos(x)

Compute cosine of x.

cosh(x)

Compute hyperbolic cosine of x.

eq(x, y)

Return 1 if x and y are equivalent, 0 otherwise.

exp(x)

Compute exponential of x (with base "e", the Euler's number).

floor(expr)

Round the value of expression expr downwards to the nearest integer. For example, ``floor(-1.5)'' is ``-2.0''.

gauss(x)

Compute Gauss function of x, corresponding to "exp(-x*x/2) / sqrt(2*PI)".

gcd(x, y)

Return the greatest common divisor of x and y. If both x and y are 0 or either or both are less than zero then behavior is undefined.

gt(x, y)

Return 1 if x is greater than y, 0 otherwise.

gte(x, y)

Return 1 if x is greater than or equal to y, 0 otherwise.

hypot(x, y)

This function is similar to the C function with the same name; it returns "sqrt(x*x + y*y)", the length of the hypotenuse of a right triangle with sides of length x and y, or the distance of the point (x, y) from the origin.

if(x, y)

Evaluate x, and if the result is non-zero return the result of the evaluation of y, return 0 otherwise.

if(x, y, z)

Evaluate x, and if the result is non-zero return the evaluation result of y, otherwise the evaluation result of z.

ifnot(x, y)

Evaluate x, and if the result is zero return the result of the evaluation of y, return 0 otherwise.

ifnot(x, y, z)

Evaluate x, and if the result is zero return the evaluation result of y, otherwise the evaluation result of z.

isinf(x)

Return 1.0 if x is +/-INFINITY, 0.0 otherwise.

isnan(x)

Return 1.0 if x is NAN, 0.0 otherwise.

ld(var)

Load the value of the internal variable with number var, which was previously stored with st(var, expr). The function returns the loaded value.

log(x)

Compute natural logarithm of x.

lt(x, y)

Return 1 if x is lesser than y, 0 otherwise.

lte(x, y)

Return 1 if x is lesser than or equal to y, 0 otherwise.

max(x, y)

Return the maximum between x and y.

min(x, y)

Return the minimum between x and y.

mod(x, y)

Compute the remainder of division of x by y.

not(expr)

Return 1.0 if expr is zero, 0.0 otherwise.

pow(x, y)

Compute the power of x elevated y, it is equivalent to "(x)^(y)".

print(t)

print(t, l)

Print the value of expression t with loglevel l. If l is not specified then a default log level is used. Returns the value of the expression printed.

Prints t with loglevel l

random(x)

Return a pseudo random value between 0.0 and 1.0. x is the index of the internal variable which will be used to save the seed/state.

root(expr, max)

Find an input value for which the function represented by expr with argument ld(0) is 0 in the interval 0..max.

The expression in expr must denote a continuous function or the result is undefined.

ld(0) is used to represent the function input value, which means that the given expression will be evaluated multiple times with various input values that the expression can access through ld(0). When the expression evaluates to 0 then the corresponding input value will be returned.

sin(x)

Compute sine of x.

sinh(x)

Compute hyperbolic sine of x.

sqrt(expr)

Compute the square root of expr. This is equivalent to "(expr)^.5".

squish(x)

Compute expression "1/(1 + exp(4*x))".

st(var, expr)

Store the value of the expression expr in an internal variable. var specifies the number of the variable where to store the value, and it is a value ranging from 0 to 9. The function returns the value stored in the internal variable. Note, Variables are currently not shared between expressions.

tan(x)

Compute tangent of x.

tanh(x)

Compute hyperbolic tangent of x.

taylor(expr, x)

taylor(expr, x, id)

Evaluate a Taylor series at x, given an expression representing the "ld(id)"-th derivative of a function at 0.

When the series does not converge the result is undefined.

ld(id) is used to represent the derivative order in expr, which means that the given expression will be evaluated multiple times with various input values that the expression can access through "ld(id)". If id is not specified then 0 is assumed.

Note, when you have the derivatives at y instead of 0, "taylor(expr, x-y)" can be used.

time(0)

Return the current (wallclock) time in seconds.

trunc(expr)

Round the value of expression expr towards zero to the nearest integer. For example, ``trunc(-1.5)'' is ``-1.0''.

while(cond, expr)

Evaluate expression expr while the expression cond is non-zero, and returns the value of the last expr evaluation, or NAN if cond was always false.

The following constants are available:

PI: area of the unit disc, approximately 3.14
E: exp(1) (Euler's number), approximately 2.718
PHI: golden ratio (1+sqrt(5))/2, approximately 1.618

Assuming that an expression is considered ``true'' if it has a non-zero value, note that:

"*" works like AND

"+" works like OR

For example the construct:

        if (A AND B) then C

is equivalent to:

        if(A*B, C)

In your C code, you can extend the list of unary and binary functions, and define recognized constants, so that they are available for your expressions.

The evaluator also recognizes the International System unit prefixes. If 'i' is appended after the prefix, binary prefixes are used, which are based on powers of 1024 instead of powers of 1000. The 'B' postfix multiplies the value by 8, and can be appended after a unit prefix or used alone. This allows using for example 'KB', 'MiB', 'G' and 'B' as number postfix.

The list of available International System prefixes follows, with indication of the corresponding powers of 10 and of 2.

y: 10^-24 / 2^-80
z: 10^-21 / 2^-70
a: 10^-18 / 2^-60
f: 10^-15 / 2^-50
p: 10^-12 / 2^-40
n: 10^-9 / 2^-30
u: 10^-6 / 2^-20
m: 10^-3 / 2^-10
c: 10^-2
d: 10^-1
h: 10^2
k: 10^3 / 2^10
K: 10^3 / 2^10
M: 10^6 / 2^20
G: 10^9 / 2^30
T: 10^12 / 2^40
P: 10^15 / 2^40
E: 10^18 / 2^50
Z: 10^21 / 2^60
Y: 10^24 / 2^70

OPENCL OPTIONS

When FFmpeg is configured with "--enable-opencl", it is possible to set the options for the global OpenCL context.

The list of supported options follows:

build_options: Set build options used to compile the registered kernels.
See reference ``OpenCL Specification Version: 1.2 chapter 5.6.4''.
platform_idx: Select the index of the platform to run OpenCL code.
The specified index must be one of the indexes in the device list which can be obtained with "ffmpeg -opencl_bench" or "av_opencl_get_device_list()".
device_idx: Select the index of the device used to run OpenCL code.
The specified index must be one of the indexes in the device list which can be obtained with "ffmpeg -opencl_bench" or "av_opencl_get_device_list()".

CODEC OPTIONS

libavcodec provides some generic global options, which can be set on all the encoders and decoders. In addition each codec may support so-called private options, which are specific for a given codec.

Sometimes, a global option may only affect a specific kind of codec, and may be nonsensical or ignored by another, so you need to be aware of the meaning of the specified options. Also some options are meant only for decoding or encoding.

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

The list of supported options follow:

b integer (encoding,audio,video)

Set bitrate in bits/s. Default value is 200K.

ab integer (encoding,audio)

Set audio bitrate (in bits/s). Default value is 128K.

bt integer (encoding,video)

Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate tolerance specifies how far ratecontrol is willing to deviate from the target average bitrate value. This is not related to min/max bitrate. Lowering tolerance too much has an adverse effect on quality.

flags flags (decoding/encoding,audio,video,subtitles)

Set generic flags.

Possible values:

mv4: Use four motion vector by macroblock (mpeg4).
qpel: Use 1/4 pel motion compensation.
loop: Use loop filter.
qscale: Use fixed qscale.
gmc: Use gmc.
mv0: Always try a mb with mv=<0,0>.
input_preserved
pass1: Use internal 2pass ratecontrol in first pass mode.
pass2: Use internal 2pass ratecontrol in second pass mode.
gray: Only decode/encode grayscale.
emu_edge: Do not draw edges.
psnr: Set error[?] variables during encoding.
truncated
naq: Normalize adaptive quantization.
ildct: Use interlaced DCT.
low_delay: Force low delay.
global_header: Place global headers in extradata instead of every keyframe.
bitexact: Only write platform-, build- and time-independent data. (except (I)DCT). This ensures that file and data checksums are reproducible and match between platforms. Its primary use is for regression testing.
aic: Apply H263 advanced intra coding / mpeg4 ac prediction.
cbp: Deprecated, use mpegvideo private options instead.
qprd: Deprecated, use mpegvideo private options instead.
ilme: Apply interlaced motion estimation.
cgop: Use closed gop.

me_method integer (encoding,video)

Set motion estimation method.

Possible values:

zero: zero motion estimation (fastest)
full: full motion estimation (slowest)
epzs: EPZS motion estimation (default)
esa: esa motion estimation (alias for full)
tesa: tesa motion estimation
dia: dia motion estimation (alias for epzs)
log: log motion estimation
phods: phods motion estimation
x1: X1 motion estimation
hex: hex motion estimation
umh: umh motion estimation
iter: iter motion estimation

extradata_size integer

Set extradata size.

time_base rational number

Set codec time base.

It is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented. For fixed-fps content, timebase should be "1 / frame_rate" and timestamp increments should be identically 1.

g integer (encoding,video)

Set the group of picture (GOP) size. Default value is 12.

ar integer (decoding/encoding,audio)

Set audio sampling rate (in Hz).

ac integer (decoding/encoding,audio)

Set number of audio channels.

cutoff integer (encoding,audio)

Set cutoff bandwidth.

frame_size integer (encoding,audio)

Set audio frame size.

Each submitted frame except the last must contain exactly frame_size samples per channel. May be 0 when the codec has CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is not restricted. It is set by some decoders to indicate constant frame size.

frame_number integer

Set the frame number.

delay integer

qcomp float (encoding,video)

Set video quantizer scale compression (VBR). It is used as a constant in the ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0.

qblur float (encoding,video)

Set video quantizer scale blur (VBR).

qmin integer (encoding,video)

Set min video quantizer scale (VBR). Must be included between -1 and 69, default value is 2.

qmax integer (encoding,video)

Set max video quantizer scale (VBR). Must be included between -1 and 1024, default value is 31.

qdiff integer (encoding,video)

Set max difference between the quantizer scale (VBR).

bf integer (encoding,video)

Set max number of B frames between non-B-frames.

Must be an integer between -1 and 16. 0 means that B-frames are disabled. If a value of -1 is used, it will choose an automatic value depending on the encoder.

Default value is 0.

b_qfactor float (encoding,video)

Set qp factor between P and B frames.

rc_strategy integer (encoding,video)

Set ratecontrol method.

b_strategy integer (encoding,video)

Set strategy to choose between I/P/B-frames.

ps integer (encoding,video)

Set RTP payload size in bytes.

mv_bits integer

header_bits integer

i_tex_bits integer

p_tex_bits integer

i_count integer

p_count integer

skip_count integer

misc_bits integer

frame_bits integer

codec_tag integer

bug flags (decoding,video)

Workaround not auto detected encoder bugs.

Possible values:

autodetect
old_msmpeg4: some old lavc generated msmpeg4v3 files (no autodetection)
xvid_ilace: Xvid interlacing bug (autodetected if fourcc==XVIX)
ump4: (autodetected if fourcc==UMP4)
no_padding: padding bug (autodetected)
amv
ac_vlc: illegal vlc bug (autodetected per fourcc)
qpel_chroma
std_qpel: old standard qpel (autodetected per fourcc/version)
qpel_chroma2
direct_blocksize: direct-qpel-blocksize bug (autodetected per fourcc/version)
edge: edge padding bug (autodetected per fourcc/version)
hpel_chroma
dc_clip
ms: Workaround various bugs in microsoft broken decoders.
trunc: trancated frames

lelim integer (encoding,video)

Set single coefficient elimination threshold for luminance (negative values also consider DC coefficient).

celim integer (encoding,video)

Set single coefficient elimination threshold for chrominance (negative values also consider dc coefficient)

strict integer (decoding/encoding,audio,video)

Specify how strictly to follow the standards.

Possible values:

very: strictly conform to an older more strict version of the spec or reference software
strict: strictly conform to all the things in the spec no matter what consequences
normal
unofficial: allow unofficial extensions
experimental: allow non standardized experimental things, experimental (unfinished/work in progress/not well tested) decoders and encoders. Note: experimental decoders can pose a security risk, do not use this for decoding untrusted input.

b_qoffset float (encoding,video)

Set QP offset between P and B frames.

err_detect flags (decoding,audio,video)

Set error detection flags.

Possible values:

crccheck: verify embedded CRCs
bitstream: detect bitstream specification deviations
buffer: detect improper bitstream length
explode: abort decoding on minor error detection
ignore_err: ignore decoding errors, and continue decoding. This is useful if you want to analyze the content of a video and thus want everything to be decoded no matter what. This option will not result in a video that is pleasing to watch in case of errors.
careful: consider things that violate the spec and have not been seen in the wild as errors
compliant: consider all spec non compliancies as errors
aggressive: consider things that a sane encoder should not do as an error

has_b_frames integer

block_align integer

mpeg_quant integer (encoding,video)

Use MPEG quantizers instead of H.263.

qsquish float (encoding,video)

How to keep quantizer between qmin and qmax (0 = clip, 1 = use differentiable function).

rc_qmod_amp float (encoding,video)

Set experimental quantizer modulation.

rc_qmod_freq integer (encoding,video)

Set experimental quantizer modulation.

rc_override_count integer

rc_eq string (encoding,video)

Set rate control equation. When computing the expression, besides the standard functions defined in the section 'Expression Evaluation', the following functions are available: bits2qp(bits), qp2bits(qp). Also the following constants are available: iTex pTex tex mv fCode iCount mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex avgTex.

maxrate integer (encoding,audio,video)

Set max bitrate tolerance (in bits/s). Requires bufsize to be set.

minrate integer (encoding,audio,video)

Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use elsewise.

bufsize integer (encoding,audio,video)

Set ratecontrol buffer size (in bits).

rc_buf_aggressivity float (encoding,video)

Currently useless.

i_qfactor float (encoding,video)

Set QP factor between P and I frames.

i_qoffset float (encoding,video)

Set QP offset between P and I frames.

rc_init_cplx float (encoding,video)

Set initial complexity for 1-pass encoding.

dct integer (encoding,video)

Set DCT algorithm.

Possible values:

auto: autoselect a good one (default)
fastint: fast integer
int: accurate integer
mmx
altivec
faan: floating point AAN DCT

lumi_mask float (encoding,video)

Compress bright areas stronger than medium ones.

tcplx_mask float (encoding,video)

Set temporal complexity masking.

scplx_mask float (encoding,video)

Set spatial complexity masking.

p_mask float (encoding,video)

Set inter masking.

dark_mask float (encoding,video)

Compress dark areas stronger than medium ones.

idct integer (decoding/encoding,video)

Select IDCT implementation.

Possible values:

auto
int
simple
simplemmx
simpleauto: Automatically pick a IDCT compatible with the simple one
arm
altivec
sh4
simplearm
simplearmv5te
simplearmv6
simpleneon
simplealpha
ipp
xvidmmx
faani: floating point AAN IDCT

slice_count integer

ec flags (decoding,video)

Set error concealment strategy.

Possible values:

guess_mvs: iterative motion vector (MV) search (slow)
deblock: use strong deblock filter for damaged MBs
favor_inter: favor predicting from the previous frame instead of the current

bits_per_coded_sample integer

pred integer (encoding,video)

Set prediction method.

Possible values:

left
plane
median

aspect rational number (encoding,video)

Set sample aspect ratio.

sar rational number (encoding,video)

Set sample aspect ratio. Alias to aspect.

debug flags (decoding/encoding,audio,video,subtitles)

Print specific debug info.

Possible values:

pict: picture info
rc: rate control
bitstream
mb_type: macroblock (MB) type
qp: per-block quantization parameter (QP)
mv: motion vector
dct_coeff
green_metadata: display complexity metadata for the upcoming frame, GoP or for a given duration.
skip
startcode
pts
er: error recognition
mmco: memory management control operations (H.264)
bugs
vis_qp: visualize quantization parameter (QP), lower QP are tinted greener
vis_mb_type: visualize block types
buffers: picture buffer allocations
thread_ops: threading operations
nomc: skip motion compensation

vismv integer (decoding,video)

Visualize motion vectors (MVs).

This option is deprecated, see the codecview filter instead.

Possible values:

pf: forward predicted MVs of P-frames
bf: forward predicted MVs of B-frames
bb: backward predicted MVs of B-frames

cmp integer (encoding,video)

Set full pel me compare function.

Possible values:

sad: sum of absolute differences, fast (default)
sse: sum of squared errors
satd: sum of absolute Hadamard transformed differences
dct: sum of absolute DCT transformed differences
psnr: sum of squared quantization errors (avoid, low quality)
bit: number of bits needed for the block
rd: rate distortion optimal, slow
zero: 0
vsad: sum of absolute vertical differences
vsse: sum of squared vertical differences
nsse: noise preserving sum of squared differences
w53: 5/3 wavelet, only used in snow
w97: 9/7 wavelet, only used in snow
dctmax
chroma

subcmp integer (encoding,video)

Set sub pel me compare function.

Possible values:

sad: sum of absolute differences, fast (default)
sse: sum of squared errors
satd: sum of absolute Hadamard transformed differences
dct: sum of absolute DCT transformed differences
psnr: sum of squared quantization errors (avoid, low quality)
bit: number of bits needed for the block
rd: rate distortion optimal, slow
zero: 0
vsad: sum of absolute vertical differences
vsse: sum of squared vertical differences
nsse: noise preserving sum of squared differences
w53: 5/3 wavelet, only used in snow
w97: 9/7 wavelet, only used in snow
dctmax
chroma

mbcmp integer (encoding,video)

Set macroblock compare function.

Possible values:

sad: sum of absolute differences, fast (default)
sse: sum of squared errors
satd: sum of absolute Hadamard transformed differences
dct: sum of absolute DCT transformed differences
psnr: sum of squared quantization errors (avoid, low quality)
bit: number of bits needed for the block
rd: rate distortion optimal, slow
zero: 0
vsad: sum of absolute vertical differences
vsse: sum of squared vertical differences
nsse: noise preserving sum of squared differences
w53: 5/3 wavelet, only used in snow
w97: 9/7 wavelet, only used in snow
dctmax
chroma

ildctcmp integer (encoding,video)

Set interlaced dct compare function.

Possible values:

sad: sum of absolute differences, fast (default)
sse: sum of squared errors
satd: sum of absolute Hadamard transformed differences
dct: sum of absolute DCT transformed differences
psnr: sum of squared quantization errors (avoid, low quality)
bit: number of bits needed for the block
rd: rate distortion optimal, slow
zero: 0
vsad: sum of absolute vertical differences
vsse: sum of squared vertical differences
nsse: noise preserving sum of squared differences
w53: 5/3 wavelet, only used in snow
w97: 9/7 wavelet, only used in snow
dctmax
chroma

dia_size integer (encoding,video)

Set diamond type & size for motion estimation.

last_pred integer (encoding,video)

Set amount of motion predictors from the previous frame.

preme integer (encoding,video)

Set pre motion estimation.

precmp integer (encoding,video)

Set pre motion estimation compare function.

Possible values:

sad: sum of absolute differences, fast (default)
sse: sum of squared errors
satd: sum of absolute Hadamard transformed differences
dct: sum of absolute DCT transformed differences
psnr: sum of squared quantization errors (avoid, low quality)
bit: number of bits needed for the block
rd: rate distortion optimal, slow
zero: 0
vsad: sum of absolute vertical differences
vsse: sum of squared vertical differences
nsse: noise preserving sum of squared differences
w53: 5/3 wavelet, only used in snow
w97: 9/7 wavelet, only used in snow
dctmax
chroma

pre_dia_size integer (encoding,video)

Set diamond type & size for motion estimation pre-pass.

subq integer (encoding,video)

Set sub pel motion estimation quality.

dtg_active_format integer

me_range integer (encoding,video)

Set limit motion vectors range (1023 for DivX player).

ibias integer (encoding,video)

Set intra quant bias.

pbias integer (encoding,video)

Set inter quant bias.

color_table_id integer

global_quality integer (encoding,audio,video)

coder integer (encoding,video)

Possible values:

vlc: variable length coder / huffman coder
ac: arithmetic coder
raw: raw (no encoding)
rle: run-length coder
deflate: deflate-based coder

context integer (encoding,video)

Set context model.

slice_flags integer

xvmc_acceleration integer

mbd integer (encoding,video)

Set macroblock decision algorithm (high quality mode).

Possible values:

simple: use mbcmp (default)
bits: use fewest bits
rd: use best rate distortion

stream_codec_tag integer

sc_threshold integer (encoding,video)

Set scene change threshold.

lmin integer (encoding,video)

Set min lagrange factor (VBR).

lmax integer (encoding,video)

Set max lagrange factor (VBR).

nr integer (encoding,video)

Set noise reduction.

rc_init_occupancy integer (encoding,video)

Set number of bits which should be loaded into the rc buffer before decoding starts.

flags2 flags (decoding/encoding,audio,video)

Possible values:

fast: Allow non spec compliant speedup tricks.
sgop: Deprecated, use mpegvideo private options instead.
noout: Skip bitstream encoding.
ignorecrop: Ignore cropping information from sps.
local_header: Place global headers at every keyframe instead of in extradata.
chunks: Frame data might be split into multiple chunks.
showall: Show all frames before the first keyframe.
skiprd: Deprecated, use mpegvideo private options instead.
export_mvs: Export motion vectors into frame side-data (see "AV_FRAME_DATA_MOTION_VECTORS") for codecs that support it. See also doc/examples/export_mvs.c.

error integer (encoding,video)

qns integer (encoding,video)

Deprecated, use mpegvideo private options instead.

threads integer (decoding/encoding,video)

Set the number of threads to be used, in case the selected codec implementation supports multi-threading.

Possible values:

auto, 0: automatically select the number of threads to set

: Default value is auto.

me_threshold integer (encoding,video)

Set motion estimation threshold.

mb_threshold integer (encoding,video)

Set macroblock threshold.

dc integer (encoding,video)

Set intra_dc_precision.

nssew integer (encoding,video)

Set nsse weight.

skip_top integer (decoding,video)

Set number of macroblock rows at the top which are skipped.

skip_bottom integer (decoding,video)

Set number of macroblock rows at the bottom which are skipped.

profile integer (encoding,audio,video)

Possible values:

unknown
aac_main
aac_low
aac_ssr
aac_ltp
aac_he
aac_he_v2
aac_ld
aac_eld
mpeg2_aac_low
mpeg2_aac_he
mpeg4_sp
mpeg4_core
mpeg4_main
mpeg4_asp
dts
dts_es
dts_96_24
dts_hd_hra
dts_hd_ma

level integer (encoding,audio,video)

Possible values:

unknown

lowres integer (decoding,audio,video)

Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.

skip_threshold integer (encoding,video)

Set frame skip threshold.

skip_factor integer (encoding,video)

Set frame skip factor.

skip_exp integer (encoding,video)

Set frame skip exponent. Negative values behave identical to the corresponding positive ones, except that the score is normalized. Positive values exist primarily for compatibility reasons and are not so useful.

skipcmp integer (encoding,video)

Set frame skip compare function.

Possible values:

sad: sum of absolute differences, fast (default)
sse: sum of squared errors
satd: sum of absolute Hadamard transformed differences
dct: sum of absolute DCT transformed differences
psnr: sum of squared quantization errors (avoid, low quality)
bit: number of bits needed for the block
rd: rate distortion optimal, slow
zero: 0
vsad: sum of absolute vertical differences
vsse: sum of squared vertical differences
nsse: noise preserving sum of squared differences
w53: 5/3 wavelet, only used in snow
w97: 9/7 wavelet, only used in snow
dctmax
chroma

border_mask float (encoding,video)

Increase the quantizer for macroblocks close to borders.

mblmin integer (encoding,video)

Set min macroblock lagrange factor (VBR).

mblmax integer (encoding,video)

Set max macroblock lagrange factor (VBR).

mepc integer (encoding,video)

Set motion estimation bitrate penalty compensation (1.0 = 256).

skip_loop_filter integer (decoding,video)

skip_idct integer (decoding,video)

skip_frame integer (decoding,video)

Make decoder discard processing depending on the frame type selected by the option value.

skip_loop_filter skips frame loop filtering, skip_idct skips frame IDCT/dequantization, skip_frame skips decoding.

Possible values:

none: Discard no frame.
default: Discard useless frames like 0-sized frames.
noref: Discard all non-reference frames.
bidir: Discard all bidirectional frames.
nokey: Discard all frames excepts keyframes.
all: Discard all frames.

: Default value is default.

bidir_refine integer (encoding,video)

Refine the two motion vectors used in bidirectional macroblocks.

brd_scale integer (encoding,video)

Downscale frames for dynamic B-frame decision.

keyint_min integer (encoding,video)

Set minimum interval between IDR-frames.

refs integer (encoding,video)

Set reference frames to consider for motion compensation.

chromaoffset integer (encoding,video)

Set chroma qp offset from luma.

trellis integer (encoding,audio,video)

Set rate-distortion optimal quantization.

sc_factor integer (encoding,video)

Set value multiplied by qscale for each frame and added to scene_change_score.

mv0_threshold integer (encoding,video)

b_sensitivity integer (encoding,video)

Adjust sensitivity of b_frame_strategy 1.

compression_level integer (encoding,audio,video)

min_prediction_order integer (encoding,audio)

max_prediction_order integer (encoding,audio)

timecode_frame_start integer (encoding,video)

Set GOP timecode frame start number, in non drop frame format.

request_channels integer (decoding,audio)

Set desired number of audio channels.

bits_per_raw_sample integer

channel_layout integer (decoding/encoding,audio)

Possible values:

request_channel_layout integer (decoding,audio)

Possible values:

rc_max_vbv_use float (encoding,video)

rc_min_vbv_use float (encoding,video)

ticks_per_frame integer (decoding/encoding,audio,video)

color_primaries integer (decoding/encoding,video)

Possible values:

bt709: BT.709
bt470m: BT.470 M
bt470bg: BT.470 BG
smpte170m: SMPTE 170 M
smpte240m: SMPTE 240 M
film: Film
bt2020: BT.2020
smpte428_1: SMPTE ST 428-1
smpte431: SMPTE 431-2
smpte432: SMPTE 432-1

color_trc integer (decoding/encoding,video)

Possible values:

bt709: BT.709
gamma22: BT.470 M
gamma28: BT.470 BG
smpte170m: SMPTE 170 M
smpte240m: SMPTE 240 M
linear: Linear
log: Log
log_sqrt: Log square root
iec61966_2_4: IEC 61966-2-4
bt1361: BT.1361
iec61966_2_1: IEC 61966-2-1
bt2020_10bit: BT.2020 - 10 bit
bt2020_12bit: BT.2020 - 12 bit
smpte2084: SMPTE ST 2084
smpte428_1: SMPTE ST 428-1
arib-std-b67: ARIB STD-B67

colorspace integer (decoding/encoding,video)

Possible values:

rgb: RGB
bt709: BT.709
fcc: FCC
bt470bg: BT.470 BG
smpte170m: SMPTE 170 M
smpte240m: SMPTE 240 M
ycocg: YCOCG
bt2020_ncl: BT.2020 NCL
bt2020_cl: BT.2020 CL
smpte2085: SMPTE 2085

color_range integer (decoding/encoding,video)

If used as input parameter, it serves as a hint to the decoder, which color_range the input has.

chroma_sample_location integer (decoding/encoding,video)

log_level_offset integer

Set the log level offset.

slices integer (encoding,video)

Number of slices, used in parallelized encoding.

thread_type flags (decoding/encoding,video)

Select which multithreading methods to use.

Use of frame will increase decoding delay by one frame per thread, so clients which cannot provide future frames should not use it.

Possible values:

slice: Decode more than one part of a single frame at once.
Multithreading using slices works only when the video was encoded with slices.
frame: Decode more than one frame at once.

: Default value is slice+frame.

audio_service_type integer (encoding,audio)

Set audio service type.

Possible values:

ma: Main Audio Service
ef: Effects
vi: Visually Impaired
hi: Hearing Impaired
di: Dialogue
co: Commentary
em: Emergency
vo: Voice Over
ka: Karaoke

request_sample_fmt sample_fmt (decoding,audio)

Set sample format audio decoders should prefer. Default value is "none".

pkt_timebase rational number

sub_charenc encoding (decoding,subtitles)

Set the input subtitles character encoding.

field_order field_order (video)

Set/override the field order of the video. Possible values:

progressive: Progressive video
tt: Interlaced video, top field coded and displayed first
bb: Interlaced video, bottom field coded and displayed first
tb: Interlaced video, top coded first, bottom displayed first
bt: Interlaced video, bottom coded first, top displayed first

skip_alpha integer (decoding,video)

Set to 1 to disable processing alpha (transparency). This works like the gray flag in the flags option which skips chroma information instead of alpha. Default is 0.

codec_whitelist list (input)

``,'' separated list of allowed decoders. By default all are allowed.

dump_separator string (input)

Separator used to separate the fields printed on the command line about the Stream parameters. For example to separate the fields with newlines and indention:

        ffprobe -dump_separator "
                                  "  -i ~/videos/matrixbench_mpeg2.mpg

DECODERS

Decoders are configured elements in FFmpeg which allow the decoding of multimedia streams.

When you configure your FFmpeg build, all the supported native decoders are enabled by default. Decoders requiring an external library must be enabled manually via the corresponding "--enable-lib" option. You can list all available decoders using the configure option "--list-decoders".

You can disable all the decoders with the configure option "--disable-decoders" and selectively enable / disable single decoders with the options "--enable-decoder=DECODER" / "--disable-decoder=DECODER".

The option "-decoders" of the ff* tools will display the list of enabled decoders.

VIDEO DECODERS

A description of some of the currently available video decoders follows.

hevc

HEVC / H.265 decoder.

Note: the skip_loop_filter option has effect only at level "all".

rawvideo

Raw video decoder.

This decoder decodes rawvideo streams.

Options

top top_field_first

Specify the assumed field type of the input video.

-1: the video is assumed to be progressive (default)
0: bottom-field-first is assumed
1: top-field-first is assumed

AUDIO DECODERS

A description of some of the currently available audio decoders follows.

ac3

AC-3 audio decoder.

This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet).

AC-3 Decoder Options

-drc_scale value

Dynamic Range Scale Factor. The factor to apply to dynamic range values from the AC-3 stream. This factor is applied exponentially. There are 3 notable scale factor ranges:

drc_scale == 0: DRC disabled. Produces full range audio.
0 < drc_scale <= 1: DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full range and full compression.
drc_scale > 1: DRC enabled. Applies drc_scale asymmetrically. Loud sounds are fully compressed. Soft sounds are enhanced.

flac

FLAC audio decoder.

This decoder aims to implement the complete FLAC specification from Xiph.

FLAC Decoder options

-use_buggy_lpc: The lavc FLAC encoder used to produce buggy streams with high lpc values (like the default value). This option makes it possible to decode such streams correctly by using lavc's old buggy lpc logic for decoding.

ffwavesynth

Internal wave synthetizer.

This decoder generates wave patterns according to predefined sequences. Its use is purely internal and the format of the data it accepts is not publicly documented.

libcelt

libcelt decoder wrapper.

libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec. Requires the presence of the libcelt headers and library during configuration. You need to explicitly configure the build with "--enable-libcelt".

libgsm

libgsm decoder wrapper.

libgsm allows libavcodec to decode the GSM full rate audio codec. Requires the presence of the libgsm headers and library during configuration. You need to explicitly configure the build with "--enable-libgsm".

This decoder supports both the ordinary GSM and the Microsoft variant.

libilbc

libilbc decoder wrapper.

libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC) audio codec. Requires the presence of the libilbc headers and library during configuration. You need to explicitly configure the build with "--enable-libilbc".

Options

The following option is supported by the libilbc wrapper.

enhance: Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled).

libopencore-amrnb

libopencore-amrnb decoder wrapper.

libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate Narrowband audio codec. Using it requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with "--enable-libopencore-amrnb".

An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB without this library.

libopencore-amrwb

libopencore-amrwb decoder wrapper.

libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate Wideband audio codec. Using it requires the presence of the libopencore-amrwb headers and library during configuration. You need to explicitly configure the build with "--enable-libopencore-amrwb".

An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB without this library.

libopus

libopus decoder wrapper.

libopus allows libavcodec to decode the Opus Interactive Audio Codec. Requires the presence of the libopus headers and library during configuration. You need to explicitly configure the build with "--enable-libopus".

An FFmpeg native decoder for Opus exists, so users can decode Opus without this library.

SUBTITLES DECODERS

dvbsub

Options

compute_clut

-1: Compute clut if no matching CLUT is in the stream.
0: Never compute CLUT
1: Always compute CLUT and override the one provided in the stream.

dvb_substream

Selects the dvb substream, or all substreams if -1 which is default.

dvdsub

This codec decodes the bitmap subtitles used in DVDs; the same subtitles can also be found in VobSub file pairs and in some Matroska files.

Options

palette: Specify the global palette used by the bitmaps. When stored in VobSub, the palette is normally specified in the index file; in Matroska, the palette is stored in the codec extra-data in the same format as in VobSub. In DVDs, the palette is stored in the IFO file, and therefore not available when reading from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal numbers (without 0x prefix) separated by comas, for example "0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b".
ifo_palette: Specify the IFO file from which the global palette is obtained. (experimental)
forced_subs_only: Only decode subtitle entries marked as forced. Some titles have forced and non-forced subtitles in the same track. Setting this flag to 1 will only keep the forced subtitles. Default value is 0.

libzvbi-teletext

Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext subtitles. Requires the presence of the libzvbi headers and library during configuration. You need to explicitly configure the build with "--enable-libzvbi".

Options

txt_page: List of teletext page numbers to decode. You may use the special * string to match all pages. Pages that do not match the specified list are dropped. Default value is *.
txt_chop_top: Discards the top teletext line. Default value is 1.
txt_format: Specifies the format of the decoded subtitles. The teletext decoder is capable of decoding the teletext pages to bitmaps or to simple text, you should use ``bitmap'' for teletext pages, because certain graphics and colors cannot be expressed in simple text. You might use ``text'' for teletext based subtitles if your application can handle simple text based subtitles. Default value is bitmap.
txt_left: X offset of generated bitmaps, default is 0.
txt_top: Y offset of generated bitmaps, default is 0.
txt_chop_spaces: Chops leading and trailing spaces and removes empty lines from the generated text. This option is useful for teletext based subtitles where empty spaces may be present at the start or at the end of the lines or empty lines may be present between the subtitle lines because of double-sized teletext charactes. Default value is 1.
txt_duration: Sets the display duration of the decoded teletext pages or subtitles in milliseconds. Default value is 30000 which is 30 seconds.
txt_transparent: Force transparent background of the generated teletext bitmaps. Default value is 0 which means an opaque background.
txt_opacity: Sets the opacity (0-255) of the teletext background. If txt_transparent is not set, it only affects characters between a start box and an end box, typically subtitles. Default value is 0 if txt_transparent is set, 255 otherwise.

ENCODERS

Encoders are configured elements in FFmpeg which allow the encoding of multimedia streams.

When you configure your FFmpeg build, all the supported native encoders are enabled by default. Encoders requiring an external library must be enabled manually via the corresponding "--enable-lib" option. You can list all available encoders using the configure option "--list-encoders".

You can disable all the encoders with the configure option "--disable-encoders" and selectively enable / disable single encoders with the options "--enable-encoder=ENCODER" / "--disable-encoder=ENCODER".

The option "-encoders" of the ff* tools will display the list of enabled encoders.

AUDIO ENCODERS

A description of some of the currently available audio encoders follows.

aac

Advanced Audio Coding (AAC) encoder.

This encoder is the default AAC encoder, natively implemented into FFmpeg. Its quality is on par or better than libfdk_aac at the default bitrate of 128kbps. This encoder also implements more options, profiles and samplerates than other encoders (with only the AAC-HE profile pending to be implemented) so this encoder has become the default and is the recommended choice.

Options

b

Set bit rate in bits/s. Setting this automatically activates constant bit rate (CBR) mode. If this option is unspecified it is set to 128kbps.

q

Set quality for variable bit rate (VBR) mode. This option is valid only using the ffmpeg command-line tool. For library interface users, use global_quality.

cutoff

Set cutoff frequency. If unspecified will allow the encoder to dynamically adjust the cutoff to improve clarity on low bitrates.

aac_coder

Set AAC encoder coding method. Possible values:

twoloop: Two loop searching (TLS) method.
This method first sets quantizers depending on band thresholds and then tries to find an optimal combination by adding or subtracting a specific value from all quantizers and adjusting some individual quantizer a little. Will tune itself based on whether aac_is, aac_ms and aac_pns are enabled. This is the default choice for a coder.
anmr: Average noise to mask ratio (ANMR) trellis-based solution.
This is an experimental coder which currently produces a lower quality, is more unstable and is slower than the default twoloop coder but has potential. Currently has no support for the aac_is or aac_pns options. Not currently recommended.
fast: Constant quantizer method.
This method sets a constant quantizer for all bands. This is the fastest of all the methods and has no rate control or support for aac_is or aac_pns. Not recommended.

aac_ms

Sets mid/side coding mode. The default value of ``auto'' will automatically use M/S with bands which will benefit from such coding. Can be forced for all bands using the value ``enable'', which is mainly useful for debugging or disabled using ``disable''.

aac_is

Sets intensity stereo coding tool usage. By default, it's enabled and will automatically toggle IS for similar pairs of stereo bands if it's benefitial. Can be disabled for debugging by setting the value to ``disable''.

aac_pns

Uses perceptual noise substitution to replace low entropy high frequency bands with imperceivable white noise during the decoding process. By default, it's enabled, but can be disabled for debugging purposes by using ``disable''.

aac_tns

Enables the use of a multitap FIR filter which spans through the high frequency bands to hide quantization noise during the encoding process and is reverted by the decoder. As well as decreasing unpleasant artifacts in the high range this also reduces the entropy in the high bands and allows for more bits to be used by the mid-low bands. By default it's enabled but can be disabled for debugging by setting the option to ``disable''.

aac_ltp

Enables the use of the long term prediction extension which increases coding efficiency in very low bandwidth situations such as encoding of voice or solo piano music by extending constant harmonic peaks in bands throughout frames. This option is implied by profile:a aac_low and is incompatible with aac_pred. Use in conjunction with -ar to decrease the samplerate.

aac_pred

Enables the use of a more traditional style of prediction where the spectral coefficients transmitted are replaced by the difference of the current coefficients minus the previous ``predicted'' coefficients. In theory and sometimes in practice this can improve quality for low to mid bitrate audio. This option implies the aac_main profile and is incompatible with aac_ltp.

profile

Sets the encoding profile, possible values:

aac_low: The default, AAC ``Low-complexity'' profile. Is the most compatible and produces decent quality.
mpeg2_aac_low: Equivalent to "-profile:a aac_low -aac_pns 0". PNS was introduced with the MPEG4 specifications.
aac_ltp: Long term prediction profile, is enabled by and will enable the aac_ltp option. Introduced in MPEG4.
aac_main: Main-type prediction profile, is enabled by and will enable the aac_pred option. Introduced in MPEG2.

: If this option is unspecified it is set to aac_low.

ac3 and ac3_fixed

AC-3 audio encoders.

These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet).

The ac3 encoder uses floating-point math, while the ac3_fixed encoder only uses fixed-point integer math. This does not mean that one is always faster, just that one or the other may be better suited to a particular system. The floating-point encoder will generally produce better quality audio for a given bitrate. The ac3_fixed encoder is not the default codec for any of the output formats, so it must be specified explicitly using the option "-acodec ac3_fixed" in order to use it.

AC-3 Metadata

The AC-3 metadata options are used to set parameters that describe the audio, but in most cases do not affect the audio encoding itself. Some of the options do directly affect or influence the decoding and playback of the resulting bitstream, while others are just for informational purposes. A few of the options will add bits to the output stream that could otherwise be used for audio data, and will thus affect the quality of the output. Those will be indicated accordingly with a note in the option list below.

These parameters are described in detail in several publicly-available documents.

*<<http://www.atsc.org/cms/standards/a_52-2010.pdf>>
*<<http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf>>

Metadata Control Options

-per_frame_metadata boolean

Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.

0: The metadata values set at initialization will be used for every frame in the stream. (default)
1: Metadata values can be changed before encoding each frame.

Downmix Levels

-center_mixlev level

Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo. This field will only be written to the bitstream if a center channel is present. The value is specified as a scale factor. There are 3 valid values:

0.707: Apply -3dB gain
0.595: Apply -4.5dB gain (default)
0.500: Apply -6dB gain

-surround_mixlev level

Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo. This field will only be written to the bitstream if one or more surround channels are present. The value is specified as a scale factor. There are 3 valid values:

0.707: Apply -3dB gain
0.500: Apply -6dB gain (default)
0.000: Silence Surround Channel(s)

Audio Production Information

Audio Production Information is optional information describing the mixing environment. Either none or both of the fields are written to the bitstream.

-mixing_level number

Mixing Level. Specifies peak sound pressure level (SPL) in the production environment when the mix was mastered. Valid values are 80 to 111, or -1 for unknown or not indicated. The default value is -1, but that value cannot be used if the Audio Production Information is written to the bitstream. Therefore, if the "room_type" option is not the default value, the "mixing_level" option must not be -1.

-room_type type

Room Type. Describes the equalization used during the final mixing session at the studio or on the dubbing stage. A large room is a dubbing stage with the industry standard X-curve equalization; a small room has flat equalization. This field will not be written to the bitstream if both the "mixing_level" option and the "room_type" option have the default values.

0
notindicated: Not Indicated (default)
1
large: Large Room
2
small: Small Room

Other Metadata Options

-copyright boolean

0
off: No Copyright Exists (default)
1
on: Copyright Exists

-dialnorm value

Dialogue Normalization. Indicates how far the average dialogue level of the program is below digital 100% full scale (0 dBFS). This parameter determines a level shift during audio reproduction that sets the average volume of the dialogue to a preset level. The goal is to match volume level between program sources. A value of -31dB will result in no volume level change, relative to the source volume, during audio reproduction. Valid values are whole numbers in the range -31 to -1, with -31 being the default.

-dsur_mode mode

Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround (Pro Logic). This field will only be written to the bitstream if the audio stream is stereo. Using this option does NOT mean the encoder will actually apply Dolby Surround processing.

0
notindicated: Not Indicated (default)
1
off: Not Dolby Surround Encoded
2
on: Dolby Surround Encoded

-original boolean

Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy.

0
off: Not Original Source
1
on: Original Source (default)

Extended Bitstream Information

The extended bitstream options are part of the Alternate Bit Stream Syntax as specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts. If any one parameter in a group is specified, all values in that group will be written to the bitstream. Default values are used for those that are written but have not been specified. If the mixing levels are written, the decoder will use these values instead of the ones specified in the "center_mixlev" and "surround_mixlev" options if it supports the Alternate Bit Stream Syntax.

Extended Bitstream Information - Part 1

-dmix_mode mode

Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt (Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.

0
notindicated: Not Indicated (default)
1
ltrt: Lt/Rt Downmix Preferred
2
loro: Lo/Ro Downmix Preferred

-ltrt_cmixlev level

Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lt/Rt mode.

1.414: Apply +3dB gain
1.189: Apply +1.5dB gain
1.000: Apply 0dB gain
0.841: Apply -1.5dB gain
0.707: Apply -3.0dB gain
0.595: Apply -4.5dB gain (default)
0.500: Apply -6.0dB gain
0.000: Silence Center Channel

-ltrt_surmixlev level

Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lt/Rt mode.

0.841: Apply -1.5dB gain
0.707: Apply -3.0dB gain
0.595: Apply -4.5dB gain
0.500: Apply -6.0dB gain (default)
0.000: Silence Surround Channel(s)

-loro_cmixlev level

Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lo/Ro mode.

1.414: Apply +3dB gain
1.189: Apply +1.5dB gain
1.000: Apply 0dB gain
0.841: Apply -1.5dB gain
0.707: Apply -3.0dB gain
0.595: Apply -4.5dB gain (default)
0.500: Apply -6.0dB gain
0.000: Silence Center Channel

-loro_surmixlev level

Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lo/Ro mode.

0.841: Apply -1.5dB gain
0.707: Apply -3.0dB gain
0.595: Apply -4.5dB gain
0.500: Apply -6.0dB gain (default)
0.000: Silence Surround Channel(s)

Extended Bitstream Information - Part 2

-dsurex_mode mode

Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX (7.1 matrixed to 5.1). Using this option does NOT mean the encoder will actually apply Dolby Surround EX processing.

0
notindicated: Not Indicated (default)
1
on: Dolby Surround EX Off
2
off: Dolby Surround EX On

-dheadphone_mode mode

Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone encoding (multi-channel matrixed to 2.0 for use with headphones). Using this option does NOT mean the encoder will actually apply Dolby Headphone processing.

0
notindicated: Not Indicated (default)
1
on: Dolby Headphone Off
2
off: Dolby Headphone On

-ad_conv_type type

A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.

0
standard: Standard A/D Converter (default)
1
hdcd: HDCD A/D Converter

Other AC-3 Encoding Options

-stereo_rematrixing boolean: Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This is an optional AC-3 feature that increases quality by selectively encoding the left/right channels as mid/side. This option is enabled by default, and it is highly recommended that it be left as enabled except for testing purposes.

Floating-Point-Only AC-3 Encoding Options

These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder due to the corresponding features not being implemented in fixed-point.

-channel_coupling boolean

Enables/Disables use of channel coupling, which is an optional AC-3 feature that increases quality by combining high frequency information from multiple channels into a single channel. The per-channel high frequency information is sent with less accuracy in both the frequency and time domains. This allows more bits to be used for lower frequencies while preserving enough information to reconstruct the high frequencies. This option is enabled by default for the floating-point encoder and should generally be left as enabled except for testing purposes or to increase encoding speed.

-1
auto: Selected by Encoder (default)
0
off: Disable Channel Coupling
1
on: Enable Channel Coupling

-cpl_start_band number

Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a value higher than the bandwidth is used, it will be reduced to 1 less than the coupling end band. If auto is used, the start band will be determined by the encoder based on the bit rate, sample rate, and channel layout. This option has no effect if channel coupling is disabled.

-1
auto: Selected by Encoder (default)

flac

FLAC (Free Lossless Audio Codec) Encoder

Options

The following options are supported by FFmpeg's flac encoder.

compression_level

Sets the compression level, which chooses defaults for many other options if they are not set explicitly.

frame_size

Sets the size of the frames in samples per channel.

lpc_coeff_precision

Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the default.

lpc_type

Sets the first stage LPC algorithm

none: LPC is not used
fixed: fixed LPC coefficients
levinson
cholesky

lpc_passes

Number of passes to use for Cholesky factorization during LPC analysis

min_partition_order

The minimum partition order

max_partition_order

The maximum partition order

prediction_order_method

estimation
2level
4level
8level
search: Bruteforce search
log

ch_mode

Channel mode

auto: The mode is chosen automatically for each frame
indep: Chanels are independently coded
left_side
right_side
mid_side

exact_rice_parameters

Chooses if rice parameters are calculated exactly or approximately. if set to 1 then they are chosen exactly, which slows the code down slightly and improves compression slightly.

multi_dim_quant

Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is applied after the first stage to finetune the coefficients. This is quite slow and slightly improves compression.

libfdk_aac

libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.

The libfdk-aac library is based on the Fraunhofer FDK AAC code from the Android project.

Requires the presence of the libfdk-aac headers and library during configuration. You need to explicitly configure the build with "--enable-libfdk-aac". The library is also incompatible with GPL, so if you allow the use of GPL, you should configure with "--enable-gpl --enable-nonfree --enable-libfdk-aac".

This encoder is considered to produce output on par or worse at 128kbps to the the native FFmpeg AAC encoder but can often produce better sounding audio at identical or lower bitrates and has support for the AAC-HE profiles.

VBR encoding, enabled through the vbr or flags +qscale options, is experimental and only works with some combinations of parameters.

Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or higher.

For more information see the fdk-aac project at <http://sourceforge.net/p/opencore-amr/fdk-aac/>.

Options

The following options are mapped on the shared FFmpeg codec options.

b

Set bit rate in bits/s. If the bitrate is not explicitly specified, it is automatically set to a suitable value depending on the selected profile.

In case VBR mode is enabled the option is ignored.

ar

Set audio sampling rate (in Hz).

channels

Set the number of audio channels.

flags +qscale

Enable fixed quality, VBR (Variable Bit Rate) mode. Note that VBR is implicitly enabled when the vbr value is positive.

cutoff

Set cutoff frequency. If not specified (or explicitly set to 0) it will use a value automatically computed by the library. Default value is 0.

profile

Set audio profile.

The following profiles are recognized:

aac_low: Low Complexity AAC (LC)
aac_he: High Efficiency AAC (HE-AAC)
aac_he_v2: High Efficiency AAC version 2 (HE-AACv2)
aac_ld: Low Delay AAC (LD)
aac_eld: Enhanced Low Delay AAC (ELD)

: If not specified it is set to aac_low.

The following are private options of the libfdk_aac encoder.

afterburner

Enable afterburner feature if set to 1, disabled if set to 0. This improves the quality but also the required processing power.

Default value is 1.

eld_sbr

Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled if set to 0.

Default value is 0.

signaling

Set SBR/PS signaling style.

It can assume one of the following values:

default: choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)
implicit: implicit backwards compatible signaling
explicit_sbr: explicit SBR, implicit PS signaling
explicit_hierarchical: explicit hierarchical signaling

: Default value is default.

latm

Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.

Default value is 0.

header_period

Set StreamMuxConfig and PCE repetition period (in frames) for sending in-band configuration buffers within LATM/LOAS transport layer.

Must be a 16-bits non-negative integer.

Default value is 0.

vbr

Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty good) and 5 is highest quality. A value of 0 will disable VBR, and CBR (Constant Bit Rate) is enabled.

Currently only the aac_low profile supports VBR encoding.

VBR modes 1-5 correspond to roughly the following average bit rates:

1: 32 kbps/channel
2: 40 kbps/channel
3: 48-56 kbps/channel
4: 64 kbps/channel
5: about 80-96 kbps/channel

: Default value is 0.

Examples

*

Use ffmpeg to convert an audio file to VBR AAC in an M4A (MP4) container:

        ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a

*

Use ffmpeg to convert an audio file to CBR 64k kbps AAC, using the High-Efficiency AAC profile:

        ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a

libmp3lame

LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper.

Requires the presence of the libmp3lame headers and library during configuration. You need to explicitly configure the build with "--enable-libmp3lame".

See libshine for a fixed-point MP3 encoder, although with a lower quality.

Options

The following options are supported by the libmp3lame wrapper. The lame-equivalent of the options are listed in parentheses.

b (-b): Set bitrate expressed in bits/s for CBR or ABR. LAME "bitrate" is expressed in kilobits/s.
q (-V): Set constant quality setting for VBR. This option is valid only using the ffmpeg command-line tool. For library interface users, use global_quality.
compression_level (-q): Set algorithm quality. Valid arguments are integers in the 0-9 range, with 0 meaning highest quality but slowest, and 9 meaning fastest while producing the worst quality.
reservoir: Enable use of bit reservoir when set to 1. Default value is 1. LAME has this enabled by default, but can be overridden by use --nores option.
joint_stereo (-m j): Enable the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo. Default value is 1.
abr (--abr): Enable the encoder to use ABR when set to 1. The lame --abr sets the target bitrate, while this options only tells FFmpeg to use ABR still relies on b to set bitrate.

libopencore-amrnb

OpenCORE Adaptive Multi-Rate Narrowband encoder.

Requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with "--enable-libopencore-amrnb --enable-version3".

This is a mono-only encoder. Officially it only supports 8000Hz sample rate, but you can override it by setting strict to unofficial or lower.

Options

b

Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.

4750
5150
5900
6700
7400
7950
10200
12200

dtx

Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).

libshine

Shine Fixed-Point MP3 encoder wrapper.

Shine is a fixed-point MP3 encoder. It has a far better performance on platforms without an FPU, e.g. armel CPUs, and some phones and tablets. However, as it is more targeted on performance than quality, it is not on par with LAME and other production-grade encoders quality-wise. Also, according to the project's homepage, this encoder may not be free of bugs as the code was written a long time ago and the project was dead for at least 5 years.

This encoder only supports stereo and mono input. This is also CBR-only.

The original project (last updated in early 2007) is at <http://sourceforge.net/projects/libshine-fxp/>. We only support the updated fork by the Savonet/Liquidsoap project at <https://github.com/savonet/shine>.

Requires the presence of the libshine headers and library during configuration. You need to explicitly configure the build with "--enable-libshine".

libtwolame

TwoLAME MP2 encoder wrapper.

Requires the presence of the libtwolame headers and library during configuration. You need to explicitly configure the build with "--enable-libtwolame".

Options

The following options are supported by the libtwolame wrapper. The twolame-equivalent options follow the FFmpeg ones and are in parentheses.

b (-b)

Set bitrate expressed in bits/s for CBR. twolame b option is expressed in kilobits/s. Default value is 128k.

q (-V)

Set quality for experimental VBR support. Maximum value range is from -50 to 50, useful range is from -10 to 10. The higher the value, the better the quality. This option is valid only using the ffmpeg command-line tool. For library interface users, use global_quality.

mode (--mode)

Set the mode of the resulting audio. Possible values:

auto: Choose mode automatically based on the input. This is the default.
stereo: Stereo
joint_stereo: Joint stereo
dual_channel: Dual channel
mono: Mono

psymodel (--psyc-mode)

Set psychoacoustic model to use in encoding. The argument must be an integer between -1 and 4, inclusive. The higher the value, the better the quality. The default value is 3.

energy_levels (--energy)

Enable energy levels extensions when set to 1. The default value is 0 (disabled).

error_protection (--protect)

Enable CRC error protection when set to 1. The default value is 0 (disabled).

copyright (--copyright)

Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).

original (--original)

Set MPEG audio original flag when set to 1. The default value is 0 (disabled).

libvo-amrwbenc

VisualOn Adaptive Multi-Rate Wideband encoder.

Requires the presence of the libvo-amrwbenc headers and library during configuration. You need to explicitly configure the build with "--enable-libvo-amrwbenc --enable-version3".

This is a mono-only encoder. Officially it only supports 16000Hz sample rate, but you can override it by setting strict to unofficial or lower.

Options

b

Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.

6600
8850
12650
14250
15850
18250
19850
23050
23850

dtx

Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).

libopus

libopus Opus Interactive Audio Codec encoder wrapper.

Requires the presence of the libopus headers and library during configuration. You need to explicitly configure the build with "--enable-libopus".

Option Mapping

Most libopus options are modelled after the opusenc utility from opus-tools. The following is an option mapping chart describing options supported by the libopus wrapper, and their opusenc-equivalent in parentheses.

b (bitrate)

Set the bit rate in bits/s. FFmpeg's b option is expressed in bits/s, while opusenc's bitrate in kilobits/s.

vbr (vbr, hard-cbr, and cvbr)

Set VBR mode. The FFmpeg vbr option has the following valid arguments, with the opusenc equivalent options in parentheses:

off (hard-cbr): Use constant bit rate encoding.
on (vbr): Use variable bit rate encoding (the default).
constrained (cvbr): Use constrained variable bit rate encoding.

compression_level (comp)

Set encoding algorithm complexity. Valid options are integers in the 0-10 range. 0 gives the fastest encodes but lower quality, while 10 gives the highest quality but slowest encoding. The default is 10.

frame_duration (framesize)

Set maximum frame size, or duration of a frame in milliseconds. The argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller frame sizes achieve lower latency but less quality at a given bitrate. Sizes greater than 20ms are only interesting at fairly low bitrates. The default is 20ms.

packet_loss (expect-loss)

Set expected packet loss percentage. The default is 0.

application (N.A.)

Set intended application type. Valid options are listed below:

voip: Favor improved speech intelligibility.
audio: Favor faithfulness to the input (the default).
lowdelay: Restrict to only the lowest delay modes.

cutoff (N.A.)

Set cutoff bandwidth in Hz. The argument must be exactly one of the following: 4000, 6000, 8000, 12000, or 20000, corresponding to narrowband, mediumband, wideband, super wideband, and fullband respectively. The default is 0 (cutoff disabled).

mapping_family (mapping_family)

Set channel mapping family to be used by the encoder. The default value of -1 uses mapping family 0 for mono and stereo inputs, and mapping family 1 otherwise. The default also disables the surround masking and LFE bandwidth optimzations in libopus, and requires that the input contains 8 channels or fewer.

Other values include 0 for mono and stereo, 1 for surround sound with masking and LFE bandwidth optimizations, and 255 for independent streams with an unspecified channel layout.

libvorbis

libvorbis encoder wrapper.

Requires the presence of the libvorbisenc headers and library during configuration. You need to explicitly configure the build with "--enable-libvorbis".

Options

The following options are supported by the libvorbis wrapper. The oggenc-equivalent of the options are listed in parentheses.

To get a more accurate and extensive documentation of the libvorbis options, consult the libvorbisenc's and oggenc's documentations. See <http://xiph.org/vorbis/>, <http://wiki.xiph.org/Vorbis-tools>, and oggenc(1).

b (-b): Set bitrate expressed in bits/s for ABR. oggenc -b is expressed in kilobits/s.
q (-q): Set constant quality setting for VBR. The value should be a float number in the range of -1.0 to 10.0. The higher the value, the better the quality. The default value is 3.0.
This option is valid only using the ffmpeg command-line tool. For library interface users, use global_quality.
cutoff (--advanced-encode-option lowpass_frequency=N): Set cutoff bandwidth in Hz, a value of 0 disables cutoff. oggenc's related option is expressed in kHz. The default value is 0 (cutoff disabled).
minrate (-m): Set minimum bitrate expressed in bits/s. oggenc -m is expressed in kilobits/s.
maxrate (-M): Set maximum bitrate expressed in bits/s. oggenc -M is expressed in kilobits/s. This only has effect on ABR mode.
iblock (--advanced-encode-option impulse_noisetune=N): Set noise floor bias for impulse blocks. The value is a float number from -15.0 to 0.0. A negative bias instructs the encoder to pay special attention to the crispness of transients in the encoded audio. The tradeoff for better transient response is a higher bitrate.

libwavpack

A wrapper providing WavPack encoding through libwavpack.

Only lossless mode using 32-bit integer samples is supported currently.

Requires the presence of the libwavpack headers and library during configuration. You need to explicitly configure the build with "--enable-libwavpack".

Note that a libavcodec-native encoder for the WavPack codec exists so users can encode audios with this codec without using this encoder. See wavpackenc.

Options

wavpack command line utility's corresponding options are listed in parentheses, if any.

frame_size (--blocksize)

Default is 32768.

compression_level

Set speed vs. compression tradeoff. Acceptable arguments are listed below:

0 (-f): Fast mode.
1: Normal (default) settings.
2 (-h): High quality.
3 (-hh): Very high quality.
4-8 (-hh -xEXTRAPROC): Same as 3, but with extra processing enabled.
4 is the same as -x2 and 8 is the same as -x6.

wavpack

WavPack lossless audio encoder.

This is a libavcodec-native WavPack encoder. There is also an encoder based on libwavpack, but there is virtually no reason to use that encoder.

VIDEO ENCODERS

A description of some of the currently available video encoders follows.

libopenh264

Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.

This encoder requires the presence of the libopenh264 headers and library during configuration. You need to explicitly configure the build with "--enable-libopenh264". The library is detected using pkg-config.

For more information about the library see <http://www.openh264.org>.

Options

The following FFmpeg global options affect the configurations of the libopenh264 encoder.

b

Set the bitrate (as a number of bits per second).

g

Set the GOP size.

maxrate

Set the max bitrate (as a number of bits per second).

flags +global_header

Set global header in the bitstream.

slices

Set the number of slices, used in parallelized encoding. Default value is 0. This is only used when slice_mode is set to fixed.

slice_mode

Set slice mode. Can assume one of the following possible values:

fixed: a fixed number of slices
rowmb: one slice per row of macroblocks
auto: automatic number of slices according to number of threads
dyn: dynamic slicing

: Default value is auto.

loopfilter

Enable loop filter, if set to 1 (automatically enabled). To disable set a value of 0.

profile

Set profile restrictions. If set to the value of main enable CABAC (set the "SEncParamExt.iEntropyCodingModeFlag" flag to 1).

max_nal_size

Set maximum NAL size in bytes.

allow_skip_frames

Allow skipping frames to hit the target bitrate if set to 1.

jpeg2000

The native jpeg 2000 encoder is lossy by default, the "-q:v" option can be used to set the encoding quality. Lossless encoding can be selected with "-pred 1".

Options

format: Can be set to either "j2k" or "jp2" (the default) that makes it possible to store non-rgb pix_fmts.

snow

Options

iterative_dia_size: dia size for the iterative motion estimation

libtheora

libtheora Theora encoder wrapper.

Requires the presence of the libtheora headers and library during configuration. You need to explicitly configure the build with "--enable-libtheora".

For more information about the libtheora project see <http://www.theora.org/>.

Options

The following global options are mapped to internal libtheora options which affect the quality and the bitrate of the encoded stream.

b

Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In case VBR (Variable Bit Rate) mode is enabled this option is ignored.

flags

Used to enable constant quality mode (VBR) encoding through the qscale flag, and to enable the "pass1" and "pass2" modes.

g

Set the GOP size.

global_quality

Set the global quality as an integer in lambda units.

Only relevant when VBR mode is enabled with "flags +qscale". The value is converted to QP units by dividing it by "FF_QP2LAMBDA", clipped in the [0 - 10] range, and then multiplied by 6.3 to get a value in the native libtheora range [0-63]. A higher value corresponds to a higher quality.

q

Enable VBR mode when set to a non-negative value, and set constant quality value as a double floating point value in QP units.

The value is clipped in the [0-10] range, and then multiplied by 6.3 to get a value in the native libtheora range [0-63].

This option is valid only using the ffmpeg command-line tool. For library interface users, use global_quality.

Examples

*

Set maximum constant quality (VBR) encoding with ffmpeg:

        ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg

*

Use ffmpeg to convert a CBR 1000 kbps Theora video stream:

        ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg

libvpx

VP8/VP9 format supported through libvpx.

Requires the presence of the libvpx headers and library during configuration. You need to explicitly configure the build with "--enable-libvpx".

Options

The following options are supported by the libvpx wrapper. The vpxenc-equivalent options or values are listed in parentheses for easy migration.

To reduce the duplication of documentation, only the private options and some others requiring special attention are documented here. For the documentation of the undocumented generic options, see the Codec Options chapter.

To get more documentation of the libvpx options, invoke the command ffmpeg -h encoder=libvpx, ffmpeg -h encoder=libvpx-vp9 or vpxenc --help. Further information is available in the libvpx API documentation.

b (target-bitrate)

Set bitrate in bits/s. Note that FFmpeg's b option is expressed in bits/s, while vpxenc's target-bitrate is in kilobits/s.

g (kf-max-dist)

keyint_min (kf-min-dist)

qmin (min-q)

qmax (max-q)

bufsize (buf-sz, buf-optimal-sz)

Set ratecontrol buffer size (in bits). Note vpxenc's options are specified in milliseconds, the libvpx wrapper converts this value as follows: "buf-sz = bufsize * 1000 / bitrate", "buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6".

rc_init_occupancy (buf-initial-sz)

Set number of bits which should be loaded into the rc buffer before decoding starts. Note vpxenc's option is specified in milliseconds, the libvpx wrapper converts this value as follows: "rc_init_occupancy * 1000 / bitrate".

undershoot-pct

Set datarate undershoot (min) percentage of the target bitrate.

overshoot-pct

Set datarate overshoot (max) percentage of the target bitrate.

skip_threshold (drop-frame)

qcomp (bias-pct)

maxrate (maxsection-pct)

Set GOP max bitrate in bits/s. Note vpxenc's option is specified as a percentage of the target bitrate, the libvpx wrapper converts this value as follows: "(maxrate * 100 / bitrate)".

minrate (minsection-pct)

Set GOP min bitrate in bits/s. Note vpxenc's option is specified as a percentage of the target bitrate, the libvpx wrapper converts this value as follows: "(minrate * 100 / bitrate)".

minrate, maxrate, b end-usage=cbr

"(minrate == maxrate == bitrate)".

crf (end-usage=cq, cq-level)

tune (tune)

psnr (psnr)
ssim (ssim)

quality, deadline (deadline)

best: Use best quality deadline. Poorly named and quite slow, this option should be avoided as it may give worse quality output than good.
good: Use good quality deadline. This is a good trade-off between speed and quality when used with the cpu-used option.
realtime: Use realtime quality deadline.

speed, cpu-used (cpu-used)

Set quality/speed ratio modifier. Higher values speed up the encode at the cost of quality.

nr (noise-sensitivity)

static-thresh

Set a change threshold on blocks below which they will be skipped by the encoder.

slices (token-parts)

Note that FFmpeg's slices option gives the total number of partitions, while vpxenc's token-parts is given as "log2(partitions)".

max-intra-rate

Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0 means unlimited.

force_key_frames

"VPX_EFLAG_FORCE_KF"

Alternate reference frame related

auto-alt-ref: Enable use of alternate reference frames (2-pass only).
arnr-max-frames: Set altref noise reduction max frame count.
arnr-type: Set altref noise reduction filter type: backward, forward, centered.
arnr-strength: Set altref noise reduction filter strength.
rc-lookahead, lag-in-frames (lag-in-frames): Set number of frames to look ahead for frametype and ratecontrol.

error-resilient

Enable error resiliency features.

VP9-specific options

lossless

Enable lossless mode.

tile-columns

Set number of tile columns to use. Note this is given as "log2(tile_columns)". For example, 8 tile columns would be requested by setting the tile-columns option to 3.

tile-rows

Set number of tile rows to use. Note this is given as "log2(tile_rows)". For example, 4 tile rows would be requested by setting the tile-rows option to 2.

frame-parallel

Enable frame parallel decodability features.

aq-mode

Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3: cyclic refresh).

colorspace color-space

Set input color space. The VP9 bitstream supports signaling the following colorspaces:

rgb sRGB
bt709 bt709
unspecified unknown
bt470bg bt601
smpte170m smpte170
smpte240m smpte240
bt2020_ncl bt2020

For more information about libvpx see: <http://www.webmproject.org/>

libwebp

libwebp WebP Image encoder wrapper

libwebp is Google's official encoder for WebP images. It can encode in either lossy or lossless mode. Lossy images are essentially a wrapper around a VP8 frame. Lossless images are a separate codec developed by Google.

Pixel Format

Currently, libwebp only supports YUV420 for lossy and RGB for lossless due to limitations of the format and libwebp. Alpha is supported for either mode. Because of API limitations, if RGB is passed in when encoding lossy or YUV is passed in for encoding lossless, the pixel format will automatically be converted using functions from libwebp. This is not ideal and is done only for convenience.

Options

-lossless boolean

Enables/Disables use of lossless mode. Default is 0.

-compression_level integer

For lossy, this is a quality/speed tradeoff. Higher values give better quality for a given size at the cost of increased encoding time. For lossless, this is a size/speed tradeoff. Higher values give smaller size at the cost of increased encoding time. More specifically, it controls the number of extra algorithms and compression tools used, and varies the combination of these tools. This maps to the method option in libwebp. The valid range is 0 to 6. Default is 4.

-qscale float

For lossy encoding, this controls image quality, 0 to 100. For lossless encoding, this controls the effort and time spent at compressing more. The default value is 75. Note that for usage via libavcodec, this option is called global_quality and must be multiplied by FF_QP2LAMBDA.

-preset type

Configuration preset. This does some automatic settings based on the general type of the image.

none: Do not use a preset.
default: Use the encoder default.
picture: Digital picture, like portrait, inner shot
photo: Outdoor photograph, with natural lighting
drawing: Hand or line drawing, with high-contrast details
icon: Small-sized colorful images
text: Text-like

libx264, libx264rgb

x264 H.264/MPEG-4 AVC encoder wrapper.

This encoder requires the presence of the libx264 headers and library during configuration. You need to explicitly configure the build with "--enable-libx264".

libx264 supports an impressive number of features, including 8x8 and 4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC entropy coding, interlacing (MBAFF), lossless mode, psy optimizations for detail retention (adaptive quantization, psy-RD, psy-trellis).

Many libx264 encoder options are mapped to FFmpeg global codec options, while unique encoder options are provided through private options. Additionally the x264opts and x264-params private options allows one to pass a list of key=value tuples as accepted by the libx264 "x264_param_parse" function.

The x264 project website is at <http://www.videolan.org/developers/x264.html>.

The libx264rgb encoder is the same as libx264, except it accepts packed RGB pixel formats as input instead of YUV.

Supported Pixel Formats

x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at x264's configure time. FFmpeg only supports one bit depth in one particular build. In other words, it is not possible to build one FFmpeg with multiple versions of x264 with different bit depths.

Options

The following options are supported by the libx264 wrapper. The x264-equivalent options or values are listed in parentheses for easy migration.

To get a more accurate and extensive documentation of the libx264 options, invoke the command x264 --full-help or consult the libx264 documentation.

b (bitrate)

Set bitrate in bits/s. Note that FFmpeg's b option is expressed in bits/s, while x264's bitrate is in kilobits/s.

bf (bframes)

g (keyint)

qmin (qpmin)

Minimum quantizer scale.

qmax (qpmax)

Maximum quantizer scale.

qdiff (qpstep)

Maximum difference between quantizer scales.

qblur (qblur)

Quantizer curve blur

qcomp (qcomp)

Quantizer curve compression factor

refs (ref)

Number of reference frames each P-frame can use. The range is from 0-16.

sc_threshold (scenecut)

Sets the threshold for the scene change detection.

trellis (trellis)

Performs Trellis quantization to increase efficiency. Enabled by default.

nr (nr)

me_range (merange)

Maximum range of the motion search in pixels.

me_method (me)

Set motion estimation method. Possible values in the decreasing order of speed:

dia (dia)
epzs (dia): Diamond search with radius 1 (fastest). epzs is an alias for dia.
hex (hex): Hexagonal search with radius 2.
umh (umh): Uneven multi-hexagon search.
esa (esa): Exhaustive search.
tesa (tesa): Hadamard exhaustive search (slowest).

subq (subme)

Sub-pixel motion estimation method.

b_strategy (b-adapt)

Adaptive B-frame placement decision algorithm. Use only on first-pass.

keyint_min (min-keyint)

Minimum GOP size.

coder

Set entropy encoder. Possible values:

ac: Enable CABAC.
vlc: Enable CAVLC and disable CABAC. It generates the same effect as x264's --no-cabac option.

cmp

Set full pixel motion estimation comparison algorithm. Possible values:

chroma: Enable chroma in motion estimation.
sad: Ignore chroma in motion estimation. It generates the same effect as x264's --no-chroma-me option.

threads (threads)

Number of encoding threads.

thread_type

Set multithreading technique. Possible values:

slice: Slice-based multithreading. It generates the same effect as x264's --sliced-threads option.
frame: Frame-based multithreading.

flags

Set encoding flags. It can be used to disable closed GOP and enable open GOP by setting it to "-cgop". The result is similar to the behavior of x264's --open-gop option.

rc_init_occupancy (vbv-init)

preset (preset)

Set the encoding preset.

tune (tune)

Set tuning of the encoding params.

profile (profile)

Set profile restrictions.

fastfirstpass

Enable fast settings when encoding first pass, when set to 1. When set to 0, it has the same effect of x264's --slow-firstpass option.

crf (crf)

Set the quality for constant quality mode.

crf_max (crf-max)

In CRF mode, prevents VBV from lowering quality beyond this point.

qp (qp)

Set constant quantization rate control method parameter.

aq-mode (aq-mode)

Set AQ method. Possible values:

none (0): Disabled.
variance (1): Variance AQ (complexity mask).
autovariance (2): Auto-variance AQ (experimental).

aq-strength (aq-strength)

Set AQ strength, reduce blocking and blurring in flat and textured areas.

psy

Use psychovisual optimizations when set to 1. When set to 0, it has the same effect as x264's --no-psy option.

psy-rd (psy-rd)

Set strength of psychovisual optimization, in psy-rd:psy-trellis format.

rc-lookahead (rc-lookahead)

Set number of frames to look ahead for frametype and ratecontrol.

weightb

Enable weighted prediction for B-frames when set to 1. When set to 0, it has the same effect as x264's --no-weightb option.

weightp (weightp)

Set weighted prediction method for P-frames. Possible values:

none (0): Disabled
simple (1): Enable only weighted refs
smart (2): Enable both weighted refs and duplicates

ssim (ssim)

Enable calculation and printing SSIM stats after the encoding.

intra-refresh (intra-refresh)

Enable the use of Periodic Intra Refresh instead of IDR frames when set to 1.

avcintra-class (class)

Configure the encoder to generate AVC-Intra. Valid values are 50,100 and 200

bluray-compat (bluray-compat)

Configure the encoder to be compatible with the bluray standard. It is a shorthand for setting ``bluray-compat=1 force-cfr=1''.

b-bias (b-bias)

Set the influence on how often B-frames are used.

b-pyramid (b-pyramid)

Set method for keeping of some B-frames as references. Possible values:

none (none): Disabled.
strict (strict): Strictly hierarchical pyramid.
normal (normal): Non-strict (not Blu-ray compatible).

mixed-refs

Enable the use of one reference per partition, as opposed to one reference per macroblock when set to 1. When set to 0, it has the same effect as x264's --no-mixed-refs option.

8x8dct

Enable adaptive spatial transform (high profile 8x8 transform) when set to 1. When set to 0, it has the same effect as x264's --no-8x8dct option.

fast-pskip

Enable early SKIP detection on P-frames when set to 1. When set to 0, it has the same effect as x264's --no-fast-pskip option.

aud (aud)

Enable use of access unit delimiters when set to 1.

mbtree

Enable use macroblock tree ratecontrol when set to 1. When set to 0, it has the same effect as x264's --no-mbtree option.

deblock (deblock)

Set loop filter parameters, in alpha:beta form.

cplxblur (cplxblur)

Set fluctuations reduction in QP (before curve compression).

partitions (partitions)

Set partitions to consider as a comma-separated list of. Possible values in the list:

p8x8: 8x8 P-frame partition.
p4x4: 4x4 P-frame partition.
b8x8: 4x4 B-frame partition.
i8x8: 8x8 I-frame partition.
i4x4: 4x4 I-frame partition. (Enabling p4x4 requires p8x8 to be enabled. Enabling i8x8 requires adaptive spatial transform (8x8dct option) to be enabled.)
none (none): Do not consider any partitions.
all (all): Consider every partition.

direct-pred (direct)

Set direct MV prediction mode. Possible values:

none (none): Disable MV prediction.
spatial (spatial): Enable spatial predicting.
temporal (temporal): Enable temporal predicting.
auto (auto): Automatically decided.

slice-max-size (slice-max-size)

Set the limit of the size of each slice in bytes. If not specified but RTP payload size (ps) is specified, that is used.

stats (stats)

Set the file name for multi-pass stats.

nal-hrd (nal-hrd)

Set signal HRD information (requires vbv-bufsize to be set). Possible values:

none (none): Disable HRD information signaling.
vbr (vbr): Variable bit rate.
cbr (cbr): Constant bit rate (not allowed in MP4 container).

x264opts (N.A.)

Set any x264 option, see x264 --fullhelp for a list.

Argument is a list of key=value couples separated by ``:''. In filter and psy-rd options that use ``:'' as a separator themselves, use ``,'' instead. They accept it as well since long ago but this is kept undocumented for some reason.

For example to specify libx264 encoding options with ffmpeg:

        ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv

a53cc boolean

Import closed captions (which must be ATSC compatible format) into output. Only the mpeg2 and h264 decoders provide these. Default is 1 (on).

x264-params (N.A.)

Override the x264 configuration using a :-separated list of key=value parameters.

This option is functionally the same as the x264opts, but is duplicated for compatibility with the Libav fork.

For example to specify libx264 encoding options with ffmpeg:

        ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\
        cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\
        no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT

Encoding ffpresets for common usages are provided so they can be used with the general presets system (e.g. passing the pre option).

libx265

x265 H.265/HEVC encoder wrapper.

This encoder requires the presence of the libx265 headers and library during configuration. You need to explicitly configure the build with --enable-libx265.

Options

preset

Set the x265 preset.

tune

Set the x265 tune parameter.

x265-params

Set x265 options using a list of key=value couples separated by ``:''. See x265 --help for a list of options.

For example to specify libx265 encoding options with -x265-params:

        ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4

libxvid

Xvid MPEG-4 Part 2 encoder wrapper.

This encoder requires the presence of the libxvidcore headers and library during configuration. You need to explicitly configure the build with "--enable-libxvid --enable-gpl".

The native "mpeg4" encoder supports the MPEG-4 Part 2 format, so users can encode to this format without this library.

Options

The following options are supported by the libxvid wrapper. Some of the following options are listed but are not documented, and correspond to shared codec options. See the Codec Options chapter for their documentation. The other shared options which are not listed have no effect for the libxvid encoder.

b

g

qmin

qmax

mpeg_quant

threads

bf

b_qfactor

b_qoffset

flags

Set specific encoding flags. Possible values:

mv4: Use four motion vector by macroblock.
aic: Enable high quality AC prediction.
gray: Only encode grayscale.
gmc: Enable the use of global motion compensation (GMC).
qpel: Enable quarter-pixel motion compensation.
cgop: Enable closed GOP.
global_header: Place global headers in extradata instead of every keyframe.

trellis

me_method

Set motion estimation method. Possible values in decreasing order of speed and increasing order of quality:

zero: Use no motion estimation (default).
phods
x1
log: Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks. x1 and log are aliases for phods.
epzs: Enable all of the things described above, plus advanced diamond zonal search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion estimation on chroma planes.
full: Enable all of the things described above, plus extended 16x16 and 8x8 blocks search.

mbd

Set macroblock decision algorithm. Possible values in the increasing order of quality:

simple: Use macroblock comparing function algorithm (default).
bits: Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks.
rd: Enable all of the things described above, plus rate distortion-based half pixel and quarter pixel refinement for 8x8 blocks, and rate distortion-based search using square pattern.

lumi_aq

Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled).

variance_aq

Enable variance adaptive quantization when set to 1. Default is 0 (disabled).

When combined with lumi_aq, the resulting quality will not be better than any of the two specified individually. In other words, the resulting quality will be the worse one of the two effects.

ssim

Set structural similarity (SSIM) displaying method. Possible values:

off

Disable displaying of SSIM information.

avg

Output average SSIM at the end of encoding to stdout. The format of showing the average SSIM is:

        Average SSIM: %f

For users who are not familiar with C, %f means a float number, or a decimal (e.g. 0.939232).

frame

Output both per-frame SSIM data during encoding and average SSIM at the end of encoding to stdout. The format of per-frame information is:

               SSIM: avg: %1.3f min: %1.3f max: %1.3f

For users who are not familiar with C, %1.3f means a float number rounded to 3 digits after the dot (e.g. 0.932).

ssim_acc

Set SSIM accuracy. Valid options are integers within the range of 0-4, while 0 gives the most accurate result and 4 computes the fastest.

mpeg2

MPEG-2 video encoder.

Options

seq_disp_ext integer

Specifies if the encoder should write a sequence_display_extension to the output.

-1
auto: Decide automatically to write it or not (this is the default) by checking if the data to be written is different from the default or unspecified values.
0
never: Never write it.
1
always: Always write it.

png

PNG image encoder.

Private options

dpi integer: Set physical density of pixels, in dots per inch, unset by default
dpm integer: Set physical density of pixels, in dots per meter, unset by default

ProRes

Apple ProRes encoder.

FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder. The used encoder can be chosen with the "-vcodec" option.

Private Options for prores-ks

profile integer

Select the ProRes profile to encode

proxy
lt
standard
hq
4444

quant_mat integer

Select quantization matrix.

auto
default
proxy
lt
standard
hq

: If set to auto, the matrix matching the profile will be picked. If not set, the matrix providing the highest quality, default, will be picked.

bits_per_mb integer

How many bits to allot for coding one macroblock. Different profiles use between 200 and 2400 bits per macroblock, the maximum is 8000.

mbs_per_slice integer

Number of macroblocks in each slice (1-8); the default value (8) should be good in almost all situations.

vendor string

Override the 4-byte vendor ID. A custom vendor ID like apl0 would claim the stream was produced by the Apple encoder.

alpha_bits integer

Specify number of bits for alpha component. Possible values are 0, 8 and 16. Use 0 to disable alpha plane coding.

Speed considerations

In the default mode of operation the encoder has to honor frame constraints (i.e. not produce frames with size bigger than requested) while still making output picture as good as possible. A frame containing a lot of small details is harder to compress and the encoder would spend more time searching for appropriate quantizers for each slice.

Setting a higher bits_per_mb limit will improve the speed.

For the fastest encoding speed set the qscale parameter (4 is the recommended value) and do not set a size constraint.

libkvazaar

Kvazaar H.265/HEVC encoder.

Requires the presence of the libkvazaar headers and library during configuration. You need to explicitly configure the build with --enable-libkvazaar.

Options

b: Set target video bitrate in bit/s and enable rate control.
kvazaar-params: Set kvazaar parameters as a list of name=value pairs separated by commas (,). See kvazaar documentation for a list of options.

QSV encoders

The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)

The ratecontrol method is selected as follows:

*

When global_quality is specified, a quality-based mode is used. Specifically this means either

-: CQP - constant quantizer scale, when the qscale codec flag is also set (the -qscale ffmpeg option).
-: LA_ICQ - intelligent constant quality with lookahead, when the look_ahead option is also set.
-: ICQ --- intelligent constant quality otherwise.

*

Otherwise, a bitrate-based mode is used. For all of those, you should specify at least the desired average bitrate with the b option.

-: LA - VBR with lookahead, when the look_ahead option is specified.
-: VCM - video conferencing mode, when the vcm option is set.
-: CBR - constant bitrate, when maxrate is specified and equal to the average bitrate.
-: VBR - variable bitrate, when maxrate is specified, but is higher than the average bitrate.
-: AVBR - average VBR mode, when maxrate is not specified. This mode is further configured by the avbr_accuracy and avbr_convergence options.

Note that depending on your system, a different mode than the one you specified may be selected by the encoder. Set the verbosity level to verbose or higher to see the actual settings used by the QSV runtime.

Additional libavcodec global options are mapped to MSDK options as follows:

*: g/gop_size -> GopPicSize
*: bf/max_b_frames+1 -> GopRefDist
*: rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB
*: slices -> NumSlice
*: refs -> NumRefFrame
*: b_strategy/b_frame_strategy -> BRefType
*: cgop/CLOSED_GOP codec flag -> GopOptFlag
*: For the CQP mode, the i_qfactor/i_qoffset and b_qfactor/b_qoffset set the difference between QPP and QPI, and QPP and QPB respectively.
*: Setting the coder option to the value vlc will make the H.264 encoder use CAVLC instead of CABAC.

vc2

SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at professional broadcasting but since it supports yuv420, yuv422 and yuv444 at 8 (limited range or full range), 10 or 12 bits, this makes it suitable for other tasks which require low overhead and low compression (like screen recording).

Options

b

Sets target video bitrate. Usually that's around 1:6 of the uncompressed video bitrate (e.g. for 1920x1080 50fps yuv422p10 that's around 400Mbps). Higher values (close to the uncompressed bitrate) turn on lossless compression mode.

field_order

Enables field coding when set (e.g. to tt - top field first) for interlaced inputs. Should increase compression with interlaced content as it splits the fields and encodes each separately.

wavelet_depth

Sets the total amount of wavelet transforms to apply, between 1 and 5 (default). Lower values reduce compression and quality. Less capable decoders may not be able to handle values of wavelet_depth over 3.

wavelet_type

Sets the transform type. Currently only 5_3 (LeGall) and 9_7 (Deslauriers-Dubuc) are implemented, with 9_7 being the one with better compression and thus is the default.

slice_width

slice_height

Sets the slice size for each slice. Larger values result in better compression. For compatibility with other more limited decoders use slice_width of 32 and slice_height of 8.

tolerance

Sets the undershoot tolerance of the rate control system in percent. This is to prevent an expensive search from being run.

qm

Sets the quantization matrix preset to use by default or when wavelet_depth is set to 5

-: default Uses the default quantization matrix from the specifications, extended with values for the fifth level. This provides a good balance between keeping detail and omitting artifacts.
-: flat Use a completely zeroed out quantization matrix. This increases PSNR but might reduce perception. Use in bogus benchmarks.
-: color Reduces detail but attempts to preserve color at extremely low bitrates.

SUBTITLES ENCODERS

dvdsub

This codec encodes the bitmap subtitle format that is used in DVDs. Typically they are stored in VOBSUB file pairs (*.idx + *.sub), and they can also be used in Matroska files.

Options

even_rows_fix: When set to 1, enable a work-around that makes the number of pixel rows even in all subtitles. This fixes a problem with some players that cut off the bottom row if the number is odd. The work-around just adds a fully transparent row if needed. The overhead is low, typically one byte per subtitle on average.
By default, this work-around is disabled.

BITSTREAM FILTERS

When you configure your FFmpeg build, all the supported bitstream filters are enabled by default. You can list all available ones using the configure option "--list-bsfs".

You can disable all the bitstream filters using the configure option "--disable-bsfs", and selectively enable any bitstream filter using the option "--enable-bsf=BSF", or you can disable a particular bitstream filter using the option "--disable-bsf=BSF".

The option "-bsfs" of the ff* tools will display the list of all the supported bitstream filters included in your build.

The ff* tools have a -bsf option applied per stream, taking a comma-separated list of filters, whose parameters follow the filter name after a '='.

        ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT

Below is a description of the currently available bitstream filters, with their parameters, if any.

aac_adtstoasc

Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration bitstream filter.

This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS header and removes the ADTS header.

This is required for example when copying an AAC stream from a raw ADTS AAC container to a FLV or a MOV/MP4 file.

chomp

Remove zero padding at the end of a packet.

dump_extra

Add extradata to the beginning of the filtered packets.

The additional argument specifies which packets should be filtered. It accepts the values:

a: add extradata to all key packets, but only if local_header is set in the flags2 codec context field
k: add extradata to all key packets
e: add extradata to all packets

If not specified it is assumed k.

For example the following ffmpeg command forces a global header (thus disabling individual packet headers) in the H.264 packets generated by the "libx264" encoder, but corrects them by adding the header stored in extradata to the key packets:

        ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts

dca_core

Extract DCA core from DTS-HD streams.

h264_mp4toannexb

Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.264 specification).

This is required by some streaming formats, typically the MPEG-2 transport stream format (``mpegts'').

For example to remux an MP4 file containing an H.264 stream to mpegts format with ffmpeg, you can use the command:

        ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts

imxdump

Modifies the bitstream to fit in MOV and to be usable by the Final Cut Pro decoder. This filter only applies to the mpeg2video codec, and is likely not needed for Final Cut Pro 7 and newer with the appropriate -tag:v.

For example, to remux 30 MB/sec NTSC IMX to MOV:

        ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov

mjpeg2jpeg

Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.

MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be extracted without loss, e.g. by

        ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg

Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for decoding. Quoting from <http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml>:

Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that ``MJPEG, or at least the MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed --- and *omitted* --- Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won't have any idea how to decompress the data. The exact table necessary is given in the OpenDML spec.''

This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images.

        ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
        exiftran -i -9 frame*.jpg
        ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

mjpega_dump_header

movsub

mp3_header_decompress

mpeg4_unpack_bframes

Unpack DivX-style packed B-frames.

DivX-style packed B-frames are not valid MPEG-4 and were only a workaround for the broken Video for Windows subsystem. They use more space, can cause minor AV sync issues, require more CPU power to decode (unless the player has some decoded picture queue to compensate the 2,0,2,0 frame per packet style) and cause trouble if copied into a standard container like mp4 or mpeg-ps/ts, because MPEG-4 decoders may not be able to decode them, since they are not valid MPEG-4.

For example to fix an AVI file containing an MPEG-4 stream with DivX-style packed B-frames using ffmpeg, you can use the command:

        ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi

noise

Damages the contents of packets without damaging the container. Can be used for fuzzing or testing error resilience/concealment.

Parameters: A numeral string, whose value is related to how often output bytes will be modified. Therefore, values below or equal to 0 are forbidden, and the lower the more frequent bytes will be modified, with 1 meaning every byte is modified.

        ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv

applies the modification to every byte.

remove_extra

FORMAT OPTIONS

The libavformat library provides some generic global options, which can be set on all the muxers and demuxers. In addition each muxer or demuxer may support so-called private options, which are specific for that component.

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

The list of supported options follows:

avioflags flags (input/output)

Possible values:

direct: Reduce buffering.

probesize integer (input)

Set probing size in bytes, i.e. the size of the data to analyze to get stream information. A higher value will enable detecting more information in case it is dispersed into the stream, but will increase latency. Must be an integer not lesser than 32. It is 5000000 by default.

packetsize integer (output)

Set packet size.

fflags flags (input/output)

Set format flags.

Possible values:

ignidx: Ignore index.
fastseek: Enable fast, but inaccurate seeks for some formats.
genpts: Generate PTS.
nofillin: Do not fill in missing values that can be exactly calculated.
noparse: Disable AVParsers, this needs "+nofillin" too.
igndts: Ignore DTS.
discardcorrupt: Discard corrupted frames.
sortdts: Try to interleave output packets by DTS.
keepside: Do not merge side data.
latm: Enable RTP MP4A-LATM payload.
nobuffer: Reduce the latency introduced by optional buffering
bitexact: Only write platform-, build- and time-independent data. This ensures that file and data checksums are reproducible and match between platforms. Its primary use is for regression testing.
shortest: Stop muxing at the end of the shortest stream. It may be needed to increase max_interleave_delta to avoid flushing the longer streams before EOF.

seek2any integer (input)

Allow seeking to non-keyframes on demuxer level when supported if set to 1. Default is 0.

analyzeduration integer (input)

Specify how many microseconds are analyzed to probe the input. A higher value will enable detecting more accurate information, but will increase latency. It defaults to 5,000,000 microseconds = 5 seconds.

cryptokey hexadecimal string (input)

Set decryption key.

indexmem integer (input)

Set max memory used for timestamp index (per stream).

rtbufsize integer (input)

Set max memory used for buffering real-time frames.

fdebug flags (input/output)

Print specific debug info.

Possible values:

ts

max_delay integer (input/output)

Set maximum muxing or demuxing delay in microseconds.

fpsprobesize integer (input)

Set number of frames used to probe fps.

audio_preload integer (output)

Set microseconds by which audio packets should be interleaved earlier.

chunk_duration integer (output)

Set microseconds for each chunk.

chunk_size integer (output)

Set size in bytes for each chunk.

err_detect, f_err_detect flags (input)

Set error detection flags. "f_err_detect" is deprecated and should be used only via the ffmpeg tool.

Possible values:

crccheck: Verify embedded CRCs.
bitstream: Detect bitstream specification deviations.
buffer: Detect improper bitstream length.
explode: Abort decoding on minor error detection.
careful: Consider things that violate the spec and have not been seen in the wild as errors.
compliant: Consider all spec non compliancies as errors.
aggressive: Consider things that a sane encoder should not do as an error.

max_interleave_delta integer (output)

Set maximum buffering duration for interleaving. The duration is expressed in microseconds, and defaults to 1000000 (1 second).

To ensure all the streams are interleaved correctly, libavformat will wait until it has at least one packet for each stream before actually writing any packets to the output file. When some streams are ``sparse'' (i.e. there are large gaps between successive packets), this can result in excessive buffering.

This field specifies the maximum difference between the timestamps of the first and the last packet in the muxing queue, above which libavformat will output a packet regardless of whether it has queued a packet for all the streams.

If set to 0, libavformat will continue buffering packets until it has a packet for each stream, regardless of the maximum timestamp difference between the buffered packets.

use_wallclock_as_timestamps integer (input)

Use wallclock as timestamps if set to 1. Default is 0.

avoid_negative_ts integer (output)

Possible values:

make_non_negative: Shift timestamps to make them non-negative. Also note that this affects only leading negative timestamps, and not non-monotonic negative timestamps.
make_zero: Shift timestamps so that the first timestamp is 0.
auto (default): Enables shifting when required by the target format.
disabled: Disables shifting of timestamp.

: When shifting is enabled, all output timestamps are shifted by the same amount. Audio, video, and subtitles desynching and relative timestamp differences are preserved compared to how they would have been without shifting.

skip_initial_bytes integer (input)

Set number of bytes to skip before reading header and frames if set to 1. Default is 0.

correct_ts_overflow integer (input)

Correct single timestamp overflows if set to 1. Default is 1.

flush_packets integer (output)

Flush the underlying I/O stream after each packet. Default 1 enables it, and has the effect of reducing the latency; 0 disables it and may slightly increase performance in some cases.

output_ts_offset offset (output)

Set the output time offset.

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

The offset is added by the muxer to the output timestamps.

Specifying a positive offset means that the corresponding streams are delayed bt the time duration specified in offset. Default value is 0 (meaning that no offset is applied).

format_whitelist list (input)

``,'' separated list of allowed demuxers. By default all are allowed.

dump_separator string (input)

Separator used to separate the fields printed on the command line about the Stream parameters. For example to separate the fields with newlines and indention:

        ffprobe -dump_separator "
                                  "  -i ~/videos/matrixbench_mpeg2.mpg

Format stream specifiers

Format stream specifiers allow selection of one or more streams that match specific properties.

Possible forms of stream specifiers are:

stream_index: Matches the stream with this index.
stream_type[:stream_index]: stream_type is one of following: 'v' for video, 'a' for audio, 's' for subtitle, 'd' for data, and 't' for attachments. If stream_index is given, then it matches the stream number stream_index of this type. Otherwise, it matches all streams of this type.
p:program_id[:stream_index]: If stream_index is given, then it matches the stream with number stream_index in the program with the id program_id. Otherwise, it matches all streams in the program.
#stream_id: Matches the stream by a format-specific ID.

The exact semantics of stream specifiers is defined by the "avformat_match_stream_specifier()" function declared in the libavformat/avformat.h header.

DEMUXERS

Demuxers are configured elements in FFmpeg that can read the multimedia streams from a particular type of file.

When you configure your FFmpeg build, all the supported demuxers are enabled by default. You can list all available ones using the configure option "--list-demuxers".

You can disable all the demuxers using the configure option "--disable-demuxers", and selectively enable a single demuxer with the option "--enable-demuxer=DEMUXER", or disable it with the option "--disable-demuxer=DEMUXER".

The option "-formats" of the ff* tools will display the list of enabled demuxers.

The description of some of the currently available demuxers follows.

aa

Audible Format 2, 3, and 4 demuxer.

This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.

applehttp

Apple HTTP Live Streaming demuxer.

This demuxer presents all AVStreams from all variant streams. The id field is set to the bitrate variant index number. By setting the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), the caller can decide which variant streams to actually receive. The total bitrate of the variant that the stream belongs to is available in a metadata key named ``variant_bitrate''.

apng

Animated Portable Network Graphics demuxer.

This demuxer is used to demux APNG files. All headers, but the PNG signature, up to (but not including) the first fcTL chunk are transmitted as extradata. Frames are then split as being all the chunks between two fcTL ones, or between the last fcTL and IEND chunks.

-ignore_loop bool: Ignore the loop variable in the file if set.
-max_fps int: Maximum framerate in frames per second (0 for no limit).
-default_fps int: Default framerate in frames per second when none is specified in the file (0 meaning as fast as possible).

asf

Advanced Systems Format demuxer.

This demuxer is used to demux ASF files and MMS network streams.

-no_resync_search bool: Do not try to resynchronize by looking for a certain optional start code.

concat

Virtual concatenation script demuxer.

This demuxer reads a list of files and other directives from a text file and demuxes them one after the other, as if all their packets had been muxed together.

The timestamps in the files are adjusted so that the first file starts at 0 and each next file starts where the previous one finishes. Note that it is done globally and may cause gaps if all streams do not have exactly the same length.

All files must have the same streams (same codecs, same time base, etc.).

The duration of each file is used to adjust the timestamps of the next file: if the duration is incorrect (because it was computed using the bit-rate or because the file is truncated, for example), it can cause artifacts. The "duration" directive can be used to override the duration stored in each file.

Syntax

The script is a text file in extended-ASCII, with one directive per line. Empty lines, leading spaces and lines starting with '#' are ignored. The following directive is recognized:

file path

Path to a file to read; special characters and spaces must be escaped with backslash or single quotes.

All subsequent file-related directives apply to that file.

ffconcat version 1.0

Identify the script type and version. It also sets the safe option to 1 if it was -1.

To make FFmpeg recognize the format automatically, this directive must appear exactly as is (no extra space or byte-order-mark) on the very first line of the script.

duration dur

Duration of the file. This information can be specified from the file; specifying it here may be more efficient or help if the information from the file is not available or accurate.

If the duration is set for all files, then it is possible to seek in the whole concatenated video.

inpoint timestamp

In point of the file. When the demuxer opens the file it instantly seeks to the specified timestamp. Seeking is done so that all streams can be presented successfully at In point.

This directive works best with intra frame codecs, because for non-intra frame ones you will usually get extra packets before the actual In point and the decoded content will most likely contain frames before In point too.

For each file, packets before the file In point will have timestamps less than the calculated start timestamp of the file (negative in case of the first file), and the duration of the files (if not specified by the "duration" directive) will be reduced based on their specified In point.

Because of potential packets before the specified In point, packet timestamps may overlap between two concatenated files.

outpoint timestamp

Out point of the file. When the demuxer reaches the specified decoding timestamp in any of the streams, it handles it as an end of file condition and skips the current and all the remaining packets from all streams.

Out point is exclusive, which means that the demuxer will not output packets with a decoding timestamp greater or equal to Out point.

This directive works best with intra frame codecs and formats where all streams are tightly interleaved. For non-intra frame codecs you will usually get additional packets with presentation timestamp after Out point therefore the decoded content will most likely contain frames after Out point too. If your streams are not tightly interleaved you may not get all the packets from all streams before Out point and you may only will be able to decode the earliest stream until Out point.

The duration of the files (if not specified by the "duration" directive) will be reduced based on their specified Out point.

file_packet_metadata key=value

Metadata of the packets of the file. The specified metadata will be set for each file packet. You can specify this directive multiple times to add multiple metadata entries.

stream

Introduce a stream in the virtual file. All subsequent stream-related directives apply to the last introduced stream. Some streams properties must be set in order to allow identifying the matching streams in the subfiles. If no streams are defined in the script, the streams from the first file are copied.

exact_stream_id id

Set the id of the stream. If this directive is given, the string with the corresponding id in the subfiles will be used. This is especially useful for MPEG-PS (VOB) files, where the order of the streams is not reliable.

Options

This demuxer accepts the following option:

safe

If set to 1, reject unsafe file paths. A file path is considered safe if it does not contain a protocol specification and is relative and all components only contain characters from the portable character set (letters, digits, period, underscore and hyphen) and have no period at the beginning of a component.

If set to 0, any file name is accepted.

The default is 1.

-1 is equivalent to 1 if the format was automatically probed and 0 otherwise.

auto_convert

If set to 1, try to perform automatic conversions on packet data to make the streams concatenable. The default is 1.

Currently, the only conversion is adding the h264_mp4toannexb bitstream filter to H.264 streams in MP4 format. This is necessary in particular if there are resolution changes.

segment_time_metadata

If set to 1, every packet will contain the lavf.concat.start_time and the lavf.concat.duration packet metadata values which are the start_time and the duration of the respective file segments in the concatenated output expressed in microseconds. The duration metadata is only set if it is known based on the concat file. The default is 0.

Examples

*

Use absolute filenames and include some comments:

        # my first filename
        file /mnt/share/file-1.wav
        # my second filename including whitespace
        file '/mnt/share/file 2.wav'
        # my third filename including whitespace plus single quote
        file '/mnt/share/file 3'\''.wav'

*

Allow for input format auto-probing, use safe filenames and set the duration of the first file:

        ffconcat version 1.0
        
        file file-1.wav
        duration 20.0
        
        file subdir/file-2.wav

flv

Adobe Flash Video Format demuxer.

This demuxer is used to demux FLV files and RTMP network streams.

-flv_metadata bool: Allocate the streams according to the onMetaData array content.

gif

Animated GIF demuxer.

It accepts the following options:

min_delay: Set the minimum valid delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 2.
max_gif_delay: Set the maximum valid delay between frames in hundredth of seconds. Range is 0 to 65535. Default value is 65535 (nearly eleven minutes), the maximum value allowed by the specification.
default_delay: Set the default delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 10.
ignore_loop: GIF files can contain information to loop a certain number of times (or infinitely). If ignore_loop is set to 1, then the loop setting from the input will be ignored and looping will not occur. If set to 0, then looping will occur and will cycle the number of times according to the GIF. Default value is 1.

For example, with the overlay filter, place an infinitely looping GIF over another video:

        ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv

Note that in the above example the shortest option for overlay filter is used to end the output video at the length of the shortest input file, which in this case is input.mp4 as the GIF in this example loops infinitely.

image2

Image file demuxer.

This demuxer reads from a list of image files specified by a pattern. The syntax and meaning of the pattern is specified by the option pattern_type.

The pattern may contain a suffix which is used to automatically determine the format of the images contained in the files.

The size, the pixel format, and the format of each image must be the same for all the files in the sequence.

This demuxer accepts the following options:

framerate

Set the frame rate for the video stream. It defaults to 25.

loop

If set to 1, loop over the input. Default value is 0.

pattern_type

Select the pattern type used to interpret the provided filename.

pattern_type accepts one of the following values.

none

Disable pattern matching, therefore the video will only contain the specified image. You should use this option if you do not want to create sequences from multiple images and your filenames may contain special pattern characters.

sequence

Select a sequence pattern type, used to specify a sequence of files indexed by sequential numbers.

A sequence pattern may contain the string ``%d'' or "%0Nd``, which specifies the position of the characters representing a sequential number in each filename matched by the pattern. If the form ''%d0Nd" is used, the string representing the number in each filename is 0-padded and N is the total number of 0-padded digits representing the number. The literal character '%' can be specified in the pattern with the string ``%%''.

If the sequence pattern contains ``%d'' or "%0Nd", the first filename of the file list specified by the pattern must contain a number inclusively contained between start_number and start_number+start_number_range-1, and all the following numbers must be sequential.

For example the pattern ``img-%03d.bmp'' will match a sequence of filenames of the form img-001.bmp, img-002.bmp, ..., img-010.bmp, etc.; the pattern ``i%%m%%g-%d.jpg'' will match a sequence of filenames of the form i%m%g-1.jpg, i%m%g-2.jpg, ..., i%m%g-10.jpg, etc.

Note that the pattern must not necessarily contain ``%d'' or "%0Nd", for example to convert a single image file img.jpeg you can employ the command:

        ffmpeg -i img.jpeg img.png

glob

Select a glob wildcard pattern type.

The pattern is interpreted like a "glob()" pattern. This is only selectable if libavformat was compiled with globbing support.

glob_sequence (deprecated, will be removed)

Select a mixed glob wildcard/sequence pattern.

If your version of libavformat was compiled with globbing support, and the provided pattern contains at least one glob meta character among "%*?[]{}" that is preceded by an unescaped ``%'', the pattern is interpreted like a "glob()" pattern, otherwise it is interpreted like a sequence pattern.

All glob special characters "%*?[]{}" must be prefixed with ``%''. To escape a literal ``%'' you shall use ``%%''.

For example the pattern "foo-%*.jpeg" will match all the filenames prefixed by ``foo-'' and terminating with ``.jpeg'', and "foo-%?%?%?.jpeg" will match all the filenames prefixed with ``foo-'', followed by a sequence of three characters, and terminating with ``.jpeg''.

This pattern type is deprecated in favor of glob and sequence.

: Default value is glob_sequence.

pixel_format

Set the pixel format of the images to read. If not specified the pixel format is guessed from the first image file in the sequence.

start_number

Set the index of the file matched by the image file pattern to start to read from. Default value is 0.

start_number_range

Set the index interval range to check when looking for the first image file in the sequence, starting from start_number. Default value is 5.

ts_from_file

If set to 1, will set frame timestamp to modification time of image file. Note that monotonity of timestamps is not provided: images go in the same order as without this option. Default value is 0. If set to 2, will set frame timestamp to the modification time of the image file in nanosecond precision.

video_size

Set the video size of the images to read. If not specified the video size is guessed from the first image file in the sequence.

Examples

*

Use ffmpeg for creating a video from the images in the file sequence img-001.jpeg, img-002.jpeg, ..., assuming an input frame rate of 10 frames per second:

        ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv

*

As above, but start by reading from a file with index 100 in the sequence:

        ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv

*

Read images matching the ``*.png'' glob pattern , that is all the files terminating with the ``.png'' suffix:

        ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv

libgme

The Game Music Emu library is a collection of video game music file emulators.

See <http://code.google.com/p/game-music-emu/> for more information.

Some files have multiple tracks. The demuxer will pick the first track by default. The track_index option can be used to select a different track. Track indexes start at 0. The demuxer exports the number of tracks as tracks meta data entry.

For very large files, the max_size option may have to be adjusted.

libopenmpt

libopenmpt based module demuxer

See <https://lib.openmpt.org/libopenmpt/> for more information.

Some files have multiple subsongs (tracks) this can be set with the subsong option.

It accepts the following options:

subsong: Set the subsong index. This can be either 'all', 'auto', or the index of the subsong. Subsong indexes start at 0. The default is 'auto'.
The default value is to let libopenmpt choose.
layout: Set the channel layout. Valid values are 1, 2, and 4 channel layouts. The default value is STEREO.
sample_rate: Set the sample rate for libopenmpt to output. Range is from 1000 to INT_MAX. The value default is 48000.

mov/mp4/3gp/QuickTime

QuickTime / MP4 demuxer.

This demuxer accepts the following options:

enable_drefs: Enable loading of external tracks, disabled by default. Enabling this can theoretically leak information in some use cases.
use_absolute_path: Allows loading of external tracks via absolute paths, disabled by default. Enabling this poses a security risk. It should only be enabled if the source is known to be non malicious.

mpegts

MPEG-2 transport stream demuxer.

This demuxer accepts the following options:

resync_size: Set size limit for looking up a new synchronization. Default value is 65536.
fix_teletext_pts: Override teletext packet PTS and DTS values with the timestamps calculated from the PCR of the first program which the teletext stream is part of and is not discarded. Default value is 1, set this option to 0 if you want your teletext packet PTS and DTS values untouched.
ts_packetsize: Output option carrying the raw packet size in bytes. Show the detected raw packet size, cannot be set by the user.
scan_all_pmts: Scan and combine all PMTs. The value is an integer with value from -1 to 1 (-1 means automatic setting, 1 means enabled, 0 means disabled). Default value is -1.

mpjpeg

MJPEG encapsulated in multi-part MIME demuxer.

This demuxer allows reading of MJPEG, where each frame is represented as a part of multipart/x-mixed-replace stream.

strict_mime_boundary: Default implementation applies a relaxed standard to multi-part MIME boundary detection, to prevent regression with numerous existing endpoints not generating a proper MIME MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check of the boundary value.

rawvideo

Raw video demuxer.

This demuxer allows one to read raw video data. Since there is no header specifying the assumed video parameters, the user must specify them in order to be able to decode the data correctly.

This demuxer accepts the following options:

framerate: Set input video frame rate. Default value is 25.
pixel_format: Set the input video pixel format. Default value is "yuv420p".
video_size: Set the input video size. This value must be specified explicitly.

For example to read a rawvideo file input.raw with ffplay, assuming a pixel format of "rgb24", a video size of "320x240", and a frame rate of 10 images per second, use the command:

        ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw

sbg

SBaGen script demuxer.

This demuxer reads the script language used by SBaGen <http://uazu.net/sbagen/> to generate binaural beats sessions. A SBG script looks like that:

        -SE
        a: 300-2.5/3 440+4.5/0
        b: 300-2.5/0 440+4.5/3
        off: -
        NOW      == a
        +0:07:00 == b
        +0:14:00 == a
        +0:21:00 == b
        +0:30:00    off

A SBG script can mix absolute and relative timestamps. If the script uses either only absolute timestamps (including the script start time) or only relative ones, then its layout is fixed, and the conversion is straightforward. On the other hand, if the script mixes both kind of timestamps, then the NOW reference for relative timestamps will be taken from the current time of day at the time the script is read, and the script layout will be frozen according to that reference. That means that if the script is directly played, the actual times will match the absolute timestamps up to the sound controller's clock accuracy, but if the user somehow pauses the playback or seeks, all times will be shifted accordingly.

tedcaptions

JSON captions used for <http://www.ted.com/>.

TED does not provide links to the captions, but they can be guessed from the page. The file tools/bookmarklets.html from the FFmpeg source tree contains a bookmarklet to expose them.

This demuxer accepts the following option:

start_time: Set the start time of the TED talk, in milliseconds. The default is 15000 (15s). It is used to sync the captions with the downloadable videos, because they include a 15s intro.

Example: convert the captions to a format most players understand:

        ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt

MUXERS

Muxers are configured elements in FFmpeg which allow writing multimedia streams to a particular type of file.

When you configure your FFmpeg build, all the supported muxers are enabled by default. You can list all available muxers using the configure option "--list-muxers".

You can disable all the muxers with the configure option "--disable-muxers" and selectively enable / disable single muxers with the options "--enable-muxer=MUXER" / "--disable-muxer=MUXER".

The option "-formats" of the ff* tools will display the list of enabled muxers.

A description of some of the currently available muxers follows.

aiff

Audio Interchange File Format muxer.

Options

It accepts the following options:

write_id3v2: Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
id3v2_version: Select ID3v2 version to write. Currently only version 3 and 4 (aka. ID3v2.3 and ID3v2.4) are supported. The default is version 4.

asf

Advanced Systems Format muxer.

Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this muxer too.

Options

It accepts the following options:

packet_size: Set the muxer packet size. By tuning this setting you may reduce data fragmentation or muxer overhead depending on your source. Default value is 3200, minimum is 100, maximum is 64k.

chromaprint

Chromaprint fingerprinter

This muxer feeds audio data to the Chromaprint library, which generates a fingerprint for the provided audio data. It takes a single signed native-endian 16-bit raw audio stream.

Options

silence_threshold

Threshold for detecting silence, ranges from 0 to 32767. -1 for default (required for use with the AcoustID service).

algorithm

Algorithm index to fingerprint with.

fp_format

Format to output the fingerprint as. Accepts the following options:

raw: Binary raw fingerprint
compressed: Binary compressed fingerprint
base64: Base64 compressed fingerprint

crc

CRC (Cyclic Redundancy Check) testing format.

This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.

The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames.

flv

Adobe Flash Video Format muxer.

This muxer accepts the following options:

flvflags flags

Possible values:

aac_seq_header_detect: Place AAC sequence header based on audio stream data.
no_sequence_end: Disable sequence end tag.

framecrc

Per-packet CRC (Cyclic Redundancy Check) testing format.

This muxer computes and prints the Adler-32 CRC for each audio and video packet. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.

The output of the muxer consists of a line for each audio and video packet of the form:

        <stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, 0x<CRC>

CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the packet.

Examples

For example to compute the CRC of the audio and video frames in INPUT, converted to raw audio and video packets, and store it in the file out.crc:

        ffmpeg -i INPUT -f framecrc out.crc

To print the information to stdout, use the command:

        ffmpeg -i INPUT -f framecrc -

With ffmpeg, you can select the output format to which the audio and video frames are encoded before computing the CRC for each packet by specifying the audio and video codec. For example, to compute the CRC of each decoded input audio frame converted to PCM unsigned 8-bit and of each decoded input video frame converted to MPEG-2 video, use the command:

        ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -

framehash

Per-packet hash testing format.

This muxer computes and prints a cryptographic hash for each audio and video packet. This can be used for packet-by-packet equality checks without having to individually do a binary comparison on each.

By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash, but the output of explicit conversions to other codecs can also be used. It uses the SHA-256 cryptographic hash function by default, but supports several other algorithms.

The output of the muxer consists of a line for each audio and video packet of the form:

        <stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, <hash>

hash is a hexadecimal number representing the computed hash for the packet.

hash algorithm: Use the cryptographic hash function specified by the string algorithm. Supported values include "MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256" (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32" and "adler32".

Examples

To compute the SHA-256 hash of the audio and video frames in INPUT, converted to raw audio and video packets, and store it in the file out.sha256:

        ffmpeg -i INPUT -f framehash out.sha256

To print the information to stdout, using the MD5 hash function, use the command:

        ffmpeg -i INPUT -f framehash -hash md5 -

framemd5

Per-packet MD5 testing format.

This is a variant of the framehash muxer. Unlike that muxer, it defaults to using the MD5 hash function.

Examples

To compute the MD5 hash of the audio and video frames in INPUT, converted to raw audio and video packets, and store it in the file out.md5:

        ffmpeg -i INPUT -f framemd5 out.md5

To print the information to stdout, use the command:

        ffmpeg -i INPUT -f framemd5 -

See also the framehash and md5 muxers.

gif

Animated GIF muxer.

It accepts the following options:

loop: Set the number of times to loop the output. Use "-1" for no loop, 0 for looping indefinitely (default).
final_delay: Force the delay (expressed in centiseconds) after the last frame. Each frame ends with a delay until the next frame. The default is "-1", which is a special value to tell the muxer to re-use the previous delay. In case of a loop, you might want to customize this value to mark a pause for instance.

For example, to encode a gif looping 10 times, with a 5 seconds delay between the loops:

        ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif

Note 1: if you wish to extract the frames into separate GIF files, you need to force the image2 muxer:

        ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"

Note 2: the GIF format has a very large time base: the delay between two frames can therefore not be smaller than one centi second.

hash

Hash testing format.

This muxer computes and prints a cryptographic hash of all the input audio and video frames. This can be used for equality checks without having to do a complete binary comparison.

By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash, but the output of explicit conversions to other codecs can also be used. Timestamps are ignored. It uses the SHA-256 cryptographic hash function by default, but supports several other algorithms.

The output of the muxer consists of a single line of the form: algo=hash, where algo is a short string representing the hash function used, and hash is a hexadecimal number representing the computed hash.

hash algorithm: Use the cryptographic hash function specified by the string algorithm. Supported values include "MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256" (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32" and "adler32".

Examples

To compute the SHA-256 hash of the input converted to raw audio and video, and store it in the file out.sha256:

        ffmpeg -i INPUT -f hash out.sha256

To print an MD5 hash to stdout use the command:

        ffmpeg -i INPUT -f hash -hash md5 -

hls

Apple HTTP Live Streaming muxer that segments MPEG-TS according to the HTTP Live Streaming (HLS) specification.

It creates a playlist file, and one or more segment files. The output filename specifies the playlist filename.

By default, the muxer creates a file for each segment produced. These files have the same name as the playlist, followed by a sequential number and a .ts extension.

For example, to convert an input file with ffmpeg:

        ffmpeg -i in.nut out.m3u8

This example will produce the playlist, out.m3u8, and segment files: out0.ts, out1.ts, out2.ts, etc.

See also the segment muxer, which provides a more generic and flexible implementation of a segmenter, and can be used to perform HLS segmentation.

Options

This muxer supports the following options:

hls_init_time seconds

Set the initial target segment length in seconds. Default value is 0. Segment will be cut on the next key frame after this time has passed on the first m3u8 list. After the initial playlist is filled ffmpeg will cut segments at duration equal to "hls_time"

hls_time seconds

Set the target segment length in seconds. Default value is 2. Segment will be cut on the next key frame after this time has passed.

hls_list_size size

Set the maximum number of playlist entries. If set to 0 the list file will contain all the segments. Default value is 5.

hls_ts_options options_list

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

hls_wrap wrap

Set the number after which the segment filename number (the number specified in each segment file) wraps. If set to 0 the number will be never wrapped. Default value is 0.

This option is useful to avoid to fill the disk with many segment files, and limits the maximum number of segment files written to disk to wrap.

start_number number

Start the playlist sequence number from number. Default value is 0.

hls_allow_cache allowcache

Explicitly set whether the client MAY \fIs0(1) or MUST NOT \fIs0(0) cache media segments.

hls_base_url baseurl

Append baseurl to every entry in the playlist. Useful to generate playlists with absolute paths.

Note that the playlist sequence number must be unique for each segment and it is not to be confused with the segment filename sequence number which can be cyclic, for example if the wrap option is specified.

hls_segment_filename filename

Set the segment filename. Unless "hls_flags single_file" is set, filename is used as a string format with the segment number:

        ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8

This example will produce the playlist, out.m3u8, and segment files: file000.ts, file001.ts, file002.ts, etc.

use_localtime

Use strftime on filename to expand the segment filename with localtime. The segment number (%d) is not available in this mode.

        ffmpeg -i in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8

This example will produce the playlist, out.m3u8, and segment files: file-20160215-1455569023.ts, file-20160215-1455569024.ts, etc.

use_localtime_mkdir

Used together with -use_localtime, it will create up to one subdirectory which is expanded in filename.

        ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8

This example will create a directory 201560215 (if it does not exist), and then produce the playlist, out.m3u8, and segment files: 201560215/file-20160215-1455569023.ts, 201560215/file-20160215-1455569024.ts, etc.

hls_key_info_file key_info_file

Use the information in key_info_file for segment encryption. The first line of key_info_file specifies the key URI written to the playlist. The key URL is used to access the encryption key during playback. The second line specifies the path to the key file used to obtain the key during the encryption process. The key file is read as a single packed array of 16 octets in binary format. The optional third line specifies the initialization vector (IV) as a hexadecimal string to be used instead of the segment sequence number (default) for encryption. Changes to key_info_file will result in segment encryption with the new key/IV and an entry in the playlist for the new key URI/IV.

Key info file format:

        <key URI>
        <key file path>
        <IV> (optional)

Example key URIs:

        http://server/file.key
        /path/to/file.key
        file.key

Example key file paths:

        file.key
        /path/to/file.key

Example IV:

        0123456789ABCDEF0123456789ABCDEF

Key info file example:

        http://server/file.key
        /path/to/file.key
        0123456789ABCDEF0123456789ABCDEF

Example shell script:

        #!/bin/sh
        BASE_URL=${1:-'.'}
        openssl rand 16 > file.key
        echo $BASE_URL/file.key > file.keyinfo
        echo file.key >> file.keyinfo
        echo $(openssl rand -hex 16) >> file.keyinfo
        ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
          -hls_key_info_file file.keyinfo out.m3u8

hls_flags single_file

If this flag is set, the muxer will store all segments in a single MPEG-TS file, and will use byte ranges in the playlist. HLS playlists generated with this way will have the version number 4. For example:

        ffmpeg -i in.nut -hls_flags single_file out.m3u8

Will produce the playlist, out.m3u8, and a single segment file, out.ts.

hls_flags delete_segments

Segment files removed from the playlist are deleted after a period of time equal to the duration of the segment plus the duration of the playlist.

hls_flags append_list

Append new segments into the end of old segment list, and remove the "#EXT-X-ENDLIST" from the old segment list.

hls_flags round_durations

Round the duration info in the playlist file segment info to integer values, instead of using floating point.

hls_flags discont_starts

Add the "#EXT-X-DISCONTINUITY" tag to the playlist, before the first segment's information.

hls_flags omit_endlist

Do not append the "EXT-X-ENDLIST" tag at the end of the playlist.

hls_flags split_by_time

Allow segments to start on frames other than keyframes. This improves behavior on some players when the time between keyframes is inconsistent, but may make things worse on others, and can cause some oddities during seeking. This flag should be used with the "hls_time" option.

hls_flags program_date_time

Generate "EXT-X-PROGRAM-DATE-TIME" tags.

hls_playlist_type event

Emit "#EXT-X-PLAYLIST-TYPE:EVENT" in the m3u8 header. Forces hls_list_size to 0; the playlist can only be appended to.

hls_playlist_type vod

Emit "#EXT-X-PLAYLIST-TYPE:VOD" in the m3u8 header. Forces hls_list_size to 0; the playlist must not change.

method

Use the given HTTP method to create the hls files.

        ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8

This example will upload all the mpegts segment files to the HTTP server using the HTTP PUT method, and update the m3u8 files every "refresh" times using the same method. Note that the HTTP server must support the given method for uploading files.

ico

ICO file muxer.

Microsoft's icon file format (ICO) has some strict limitations that should be noted:

*

Size cannot exceed 256 pixels in any dimension

*

Only BMP and PNG images can be stored

*

If a BMP image is used, it must be one of the following pixel formats:

        BMP Bit Depth      FFmpeg Pixel Format
        1bit               pal8
        4bit               pal8
        8bit               pal8
        16bit              rgb555le
        24bit              bgr24
        32bit              bgra

*

If a BMP image is used, it must use the BITMAPINFOHEADER DIB header

*

If a PNG image is used, it must use the rgba pixel format

image2

Image file muxer.

The image file muxer writes video frames to image files.

The output filenames are specified by a pattern, which can be used to produce sequentially numbered series of files. The pattern may contain the string ``%d'' or "%0Nd``, this string specifies the position of the characters representing a numbering in the filenames. If the form ''%0Nd" is used, the string representing the number in each filename is 0-padded to N digits. The literal character '%' can be specified in the pattern with the string ``%%''.

If the pattern contains ``%d'' or "%0Nd", the first filename of the file list specified will contain the number 1, all the following numbers will be sequential.

The pattern may contain a suffix which is used to automatically determine the format of the image files to write.

For example the pattern ``img-%03d.bmp'' will specify a sequence of filenames of the form img-001.bmp, img-002.bmp, ..., img-010.bmp, etc. The pattern ``img%%-%d.jpg'' will specify a sequence of filenames of the form img%-1.jpg, img%-2.jpg, ..., img%-10.jpg, etc.

Examples

The following example shows how to use ffmpeg for creating a sequence of files img-001.jpeg, img-002.jpeg, ..., taking one image every second from the input video:

        ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg'

Note that with ffmpeg, if the format is not specified with the "-f" option and the output filename specifies an image file format, the image2 muxer is automatically selected, so the previous command can be written as:

        ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg'

Note also that the pattern must not necessarily contain ``%d'' or "%0Nd", for example to create a single image file img.jpeg from the input video you can employ the command:

        ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg

The strftime option allows you to expand the filename with date and time information. Check the documentation of the "strftime()" function for the syntax.

For example to generate image files from the "strftime()" ``%Y-%m-%d_%H-%M-%S'' pattern, the following ffmpeg command can be used:

        ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"

Options

start_number: Start the sequence from the specified number. Default value is 0.
update: If set to 1, the filename will always be interpreted as just a filename, not a pattern, and the corresponding file will be continuously overwritten with new images. Default value is 0.
strftime: If set to 1, expand the filename with date and time information from "strftime()". Default value is 0.

The image muxer supports the .Y.U.V image file format. This format is special in that that each image frame consists of three files, for each of the YUV420P components. To read or write this image file format, specify the name of the '.Y' file. The muxer will automatically open the '.U' and '.V' files as required.

matroska

Matroska container muxer.

This muxer implements the matroska and webm container specs.

Metadata

The recognized metadata settings in this muxer are:

title

Set title name provided to a single track.

language

Specify the language of the track in the Matroska languages form.

The language can be either the 3 letters bibliographic ISO-639-2 (ISO 639-2/B) form (like ``fre'' for French), or a language code mixed with a country code for specialities in languages (like ``fre-ca'' for Canadian French).

stereo_mode

Set stereo 3D video layout of two views in a single video track.

The following values are recognized:

mono: video is not stereo
left_right: Both views are arranged side by side, Left-eye view is on the left
bottom_top: Both views are arranged in top-bottom orientation, Left-eye view is at bottom
top_bottom: Both views are arranged in top-bottom orientation, Left-eye view is on top
checkerboard_rl: Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
checkerboard_lr: Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
row_interleaved_rl: Each view is constituted by a row based interleaving, Right-eye view is first row
row_interleaved_lr: Each view is constituted by a row based interleaving, Left-eye view is first row
col_interleaved_rl: Both views are arranged in a column based interleaving manner, Right-eye view is first column
col_interleaved_lr: Both views are arranged in a column based interleaving manner, Left-eye view is first column
anaglyph_cyan_red: All frames are in anaglyph format viewable through red-cyan filters
right_left: Both views are arranged side by side, Right-eye view is on the left
anaglyph_green_magenta: All frames are in anaglyph format viewable through green-magenta filters
block_lr: Both eyes laced in one Block, Left-eye view is first
block_rl: Both eyes laced in one Block, Right-eye view is first

For example a 3D WebM clip can be created using the following command line:

        ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm

Options

This muxer supports the following options:

reserve_index_space

By default, this muxer writes the index for seeking (called cues in Matroska terms) at the end of the file, because it cannot know in advance how much space to leave for the index at the beginning of the file. However for some use cases --- e.g. streaming where seeking is possible but slow --- it is useful to put the index at the beginning of the file.

If this option is set to a non-zero value, the muxer will reserve a given amount of space in the file header and then try to write the cues there when the muxing finishes. If the available space does not suffice, muxing will fail. A safe size for most use cases should be about 50kB per hour of video.

Note that cues are only written if the output is seekable and this option will have no effect if it is not.

md5

MD5 testing format.

This is a variant of the hash muxer. Unlike that muxer, it defaults to using the MD5 hash function.

Examples

To compute the MD5 hash of the input converted to raw audio and video, and store it in the file out.md5:

        ffmpeg -i INPUT -f md5 out.md5

You can print the MD5 to stdout with the command:

        ffmpeg -i INPUT -f md5 -

See also the hash and framemd5 muxers.

mov, mp4, ismv

MOV/MP4/ISMV (Smooth Streaming) muxer.

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

Options

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

-moov_size bytes: Reserves space for the moov atom at the beginning of the file instead of placing the moov atom at the end. If the space reserved is insufficient, muxing will fail.
-movflags frag_keyframe: Start a new fragment at each video keyframe.
-frag_duration duration: Create fragments that are duration microseconds long.
-frag_size size: Create fragments that contain up to size bytes of payload data.
-movflags frag_custom: Allow the caller to manually choose when to cut fragments, by calling "av_write_frame(ctx, NULL)" to write a fragment with the packets written so far. (This is only useful with other applications integrating libavformat, not from ffmpeg.)
-min_frag_duration duration: Don't create fragments that are shorter than duration microseconds long.

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

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

-movflags empty_moov: Write an initial moov atom directly at the start of the file, without describing any samples in it. Generally, an mdat/moov pair is written at the start of the file, as a normal MOV/MP4 file, containing only a short portion of the file. With this option set, there is no initial mdat atom, and the moov atom only describes the tracks but has a zero duration.
This option is implicitly set when writing ismv (Smooth Streaming) files.
-movflags separate_moof: Write a separate moof (movie fragment) atom for each track. Normally, packets for all tracks are written in a moof atom (which is slightly more efficient), but with this option set, the muxer writes one moof/mdat pair for each track, making it easier to separate tracks.
This option is implicitly set when writing ismv (Smooth Streaming) files.
-movflags faststart: Run a second pass moving the index (moov atom) to the beginning of the file. This operation can take a while, and will not work in various situations such as fragmented output, thus it is not enabled by default.
-movflags rtphint: Add RTP hinting tracks to the output file.
-movflags disable_chpl: Disable Nero chapter markers (chpl atom). Normally, both Nero chapters and a QuickTime chapter track are written to the file. With this option set, only the QuickTime chapter track will be written. Nero chapters can cause failures when the file is reprocessed with certain tagging programs, like mp3Tag 2.61a and iTunes 11.3, most likely other versions are affected as well.
-movflags omit_tfhd_offset: Do not write any absolute base_data_offset in tfhd atoms. This avoids tying fragments to absolute byte positions in the file/streams.
-movflags default_base_moof: Similarly to the omit_tfhd_offset, this flag avoids writing the absolute base_data_offset field in tfhd atoms, but does so by using the new default-base-is-moof flag instead. This flag is new from 14496-12:2012. This may make the fragments easier to parse in certain circumstances (avoiding basing track fragment location calculations on the implicit end of the previous track fragment).
-write_tmcd: Specify "on" to force writing a timecode track, "off" to disable it and "auto" to write a timecode track only for mov and mp4 output (default).

Example

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

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

Audible AAX

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

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

mp3

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

*

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

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

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

*

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

*

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

Examples:

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

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

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

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

Write a ``clean'' MP3 without any extra features:

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

mpegts

MPEG transport stream muxer.

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

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

Options

The muxer options are:

mpegts_original_network_id number

Set the original_network_id (default 0x0001). This is unique identifier of a network in DVB. Its main use is in the unique identification of a service through the path Original_Network_ID, Transport_Stream_ID.

mpegts_transport_stream_id number

Set the transport_stream_id (default 0x0001). This identifies a transponder in DVB.

mpegts_service_id number

Set the service_id (default 0x0001) also known as program in DVB.

mpegts_service_type number

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

mpegts_pmt_start_pid number

Set the first PID for PMT (default 0x1000, max 0x1f00).

mpegts_start_pid number

Set the first PID for data packets (default 0x0100, max 0x0f00).

mpegts_m2ts_mode number

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

muxrate number

Set a constant muxrate (default VBR).

pcr_period numer

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

pat_period number

Maximal time in seconds between PAT/PMT tables.

sdt_period number

Maximal time in seconds between SDT tables.

pes_payload_size number

Set minimum PES packet payload in bytes.

mpegts_flags flags

Set flags (see below).

mpegts_copyts number

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

tables_version number

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

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

Option mpegts_service_type accepts the following values:

hex_value: Any hexdecimal value between 0x01 to 0xff as defined in ETSI 300 468.
digital_tv: Digital TV service.
digital_radio: Digital Radio service.
teletext: Teletext service.
advanced_codec_digital_radio: Advanced Codec Digital Radio service.
mpeg2_digital_hdtv: MPEG2 Digital HDTV service.
advanced_codec_digital_sdtv: Advanced Codec Digital SDTV service.
advanced_codec_digital_hdtv: Advanced Codec Digital HDTV service.

Option mpegts_flags may take a set of such flags:

resend_headers: Reemit PAT/PMT before writing the next packet.
latm: Use LATM packetization for AAC.
pat_pmt_at_frames: Reemit PAT and PMT at each video frame.
system_b: Conform to System B (DVB) instead of System A (ATSC).

Example

        ffmpeg -i file.mpg -c copy \
             -mpegts_original_network_id 0x1122 \
             -mpegts_transport_stream_id 0x3344 \
             -mpegts_service_id 0x5566 \
             -mpegts_pmt_start_pid 0x1500 \
             -mpegts_start_pid 0x150 \
             -metadata service_provider="Some provider" \
             -metadata service_name="Some Channel" \
             -y out.ts

mxf, mxf_d10

MXF muxer.

Options

The muxer options are:

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

null

Null muxer.

This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes.

For example to benchmark decoding with ffmpeg you can use the command:

        ffmpeg -benchmark -i INPUT -f null out.null

Note that the above command does not read or write the out.null file, but specifying the output file is required by the ffmpeg syntax.

Alternatively you can write the command as:

        ffmpeg -benchmark -i INPUT -f null -

nut

-syncpoints flags

Change the syncpoint usage in nut:

default use the normal low-overhead seeking aids.

none do not use the syncpoints at all, reducing the overhead but making the stream non-seekable;

    Use of this option is not recommended, as the resulting files are very damage
    sensitive and seeking is not possible. Also in general the overhead from
    syncpoints is negligible. Note, -C<write_index> 0 can be used to disable
    all growing data tables, allowing to mux endless streams with limited memory
    and without these disadvantages.

timestamped extend the syncpoint with a wallclock field.

: The none and timestamped flags are experimental.

-write_index bool

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

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

ogg

Ogg container muxer.

-page_duration duration: Preferred page duration, in microseconds. The muxer will attempt to create pages that are approximately duration microseconds long. This allows the user to compromise between seek granularity and container overhead. The default is 1 second. A value of 0 will fill all segments, making pages as large as possible. A value of 1 will effectively use 1 packet-per-page in most situations, giving a small seek granularity at the cost of additional container overhead.
-serial_offset value: Serial value from which to set the streams serial number. Setting it to different and sufficiently large values ensures that the produced ogg files can be safely chained.

segment, stream_segment, ssegment

Basic stream segmenter.

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

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

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

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

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

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

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

Options

The segment muxer supports the following options:

increment_tc 1|0

if set to 1, increment timecode between each segment If this is selected, the input need to have a timecode in the first video stream. Default value is 0.

reference_stream specifier

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

segment_format format

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

segment_format_options options_list

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

segment_list name

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

segment_list_flags flags

Set flags affecting the segment list generation.

It currently supports the following flags:

cache: Allow caching (only affects M3U8 list files).
live: Allow live-friendly file generation.

segment_list_size size

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

segment_list_entry_prefix prefix

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

segment_list_type type

Select the listing format.

The following values are recognized:

flat

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

csv, ext

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

        <segment_filename>,<segment_start_time>,<segment_end_time>

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

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

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

ext is deprecated in favor or csv.

ffconcat

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

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

m3u8

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

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

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

segment_time time

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

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

segment_atclocktime 1|0

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

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

Default value is ``0''.

segment_clocktime_offset duration

Delay the segment splitting times with the specified duration when using segment_atclocktime.

For example with segment_time set to ``900'' and segment_clocktime_offset set to ``300'' this makes it possible to create files at 12:05, 12:20, 12:35, etc.

Default value is ``0''.

segment_clocktime_wrap_duration duration

Force the segmenter to only start a new segment if a packet reaches the muxer within the specified duration after the segmenting clock time. This way you can make the segmenter more resilient to backward local time jumps, such as leap seconds or transition to standard time from daylight savings time.

Assuming that the delay between the packets of your source is less than 0.5 second you can detect a leap second by specifying 0.5 as the duration.

Default is the maximum possible duration which means starting a new segment regardless of the elapsed time since the last clock time.

segment_time_delta delta

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

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

        PTS >= start_time - time_delta

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

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

segment_times times

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

segment_frames frames

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

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

segment_wrap limit

Wrap around segment index once it reaches limit.

segment_start_number number

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

strftime 1|0

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

break_non_keyframes 1|0

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

reset_timestamps 1|0

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

initial_offset offset

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

write_empty_segments 1|0

If enabled, write an empty segment if there are no packets during the period a segment would usually span. Otherwise, the segment will be filled with the next packet written. Defaults to 0.

Examples

*

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

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

*

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

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

*

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

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

*

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

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

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

*

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

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

*

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

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

*

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

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

smoothstreaming

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

window_size: Specify the number of fragments kept in the manifest. Default 0 (keep all).
extra_window_size: Specify the number of fragments kept outside of the manifest before removing from disk. Default 5.
lookahead_count: Specify the number of lookahead fragments. Default 2.
min_frag_duration: Specify the minimum fragment duration (in microseconds). Default 5000000.
remove_at_exit: Specify whether to remove all fragments when finished. Default 0 (do not remove).

fifo

The fifo pseudo-muxer allows the separation of encoding and muxing by using first-in-first-out queue and running the actual muxer in a separate thread. This is especially useful in combination with the tee muxer and can be used to send data to several destinations with different reliability/writing speed/latency.

API users should be aware that callback functions (interrupt_callback, io_open and io_close) used within its AVFormatContext must be thread-safe.

The behavior of the fifo muxer if the queue fills up or if the output fails is selectable,

*: output can be transparently restarted with configurable delay between retries based on real time or time of the processed stream.
*: encoding can be blocked during temporary failure, or continue transparently dropping packets in case fifo queue fills up.
fifo_format: Specify the format name. Useful if it cannot be guessed from the output name suffix.
queue_size: Specify size of the queue (number of packets). Default value is 60.
format_opts: Specify format options for the underlying muxer. Muxer options can be specified as a list of key=value pairs separated by ':'.
drop_pkts_on_overflow bool: If set to 1 (true), in case the fifo queue fills up, packets will be dropped rather than blocking the encoder. This makes it possible to continue streaming without delaying the input, at the cost of omitting part of the stream. By default this option is set to 0 (false), so in such cases the encoder will be blocked until the muxer processes some of the packets and none of them is lost.
attempt_recovery bool: If failure occurs, attempt to recover the output. This is especially useful when used with network output, since it makes it possible to restart streaming transparently. By default this option is set to 0 (false).
max_recovery_attempts: Sets maximum number of successive unsuccessful recovery attempts after which the output fails permanently. By default this option is set to 0 (unlimited).
recovery_wait_time duration: Waiting time before the next recovery attempt after previous unsuccessful recovery attempt. Default value is 5 seconds.
recovery_wait_streamtime bool: If set to 0 (false), the real time is used when waiting for the recovery attempt (i.e. the recovery will be attempted after at least recovery_wait_time seconds). If set to 1 (true), the time of the processed stream is taken into account instead (i.e. the recovery will be attempted after at least recovery_wait_time seconds of the stream is omitted). By default, this option is set to 0 (false).
recover_any_error bool: If set to 1 (true), recovery will be attempted regardless of type of the error causing the failure. By default this option is set to 0 (false) and in case of certain (usually permanent) errors the recovery is not attempted even when attempt_recovery is set to 1.
restart_with_keyframe bool: Specify whether to wait for the keyframe after recovering from queue overflow or failure. This option is set to 0 (false) by default.

Examples

*

Stream something to rtmp server, continue processing the stream at real-time rate even in case of temporary failure (network outage) and attempt to recover streaming every second indefinitely.

        ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a
          -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name

tee

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

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

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

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

The following special options are also recognized:

f

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

bsfs[/spec]

Specify a list of bitstream filters to apply to the specified output.

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

Several bitstream filters can be specified, separated by ``,''.

select

Select the streams that should be mapped to the slave output, specified by a stream specifier. If not specified, this defaults to all the input streams. 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 (the streams need to be explicitly mapped):

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

*

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 -strict experimental
               -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"

*

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

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

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

webm_dash_manifest

WebM DASH Manifest muxer.

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

For more information see:

*: WebM DASH Specification: <https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>
*: ISO DASH Specification: <http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>

Options

This muxer supports the following options:

adaptation_sets: This option has the following syntax: ``id=x,streams=a,b,c id=y,streams=d,e'' where x and y are the unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding audio and video streams. Any number of adaptation sets can be added using this option.
live: Set this to 1 to create a live stream DASH Manifest. Default: 0.
chunk_start_index: Start index of the first chunk. This will go in the startNumber attribute of the SegmentTemplate element in the manifest. Default: 0.
chunk_duration_ms: Duration of each chunk in milliseconds. This will go in the duration attribute of the SegmentTemplate element in the manifest. Default: 1000.
utc_timing_url: URL of the page that will return the UTC timestamp in ISO format. This will go in the value attribute of the UTCTiming element in the manifest. Default: None.
time_shift_buffer_depth: Smallest time (in seconds) shifting buffer for which any Representation is guaranteed to be available. This will go in the timeShiftBufferDepth attribute of the MPD element. Default: 60.
minimum_update_period: Minimum update period (in seconds) of the manifest. This will go in the minimumUpdatePeriod attribute of the MPD element. Default: 0.

Example

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

webm_chunk

WebM Live Chunk Muxer.

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

Options

This muxer supports the following options:

chunk_start_index: Index of the first chunk (defaults to 0).
header: Filename of the header where the initialization data will be written.
audio_chunk_duration: Duration of each audio chunk in milliseconds (defaults to 5000).

Example

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

METADATA

FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded INI-like text file and then load it back using the metadata muxer/demuxer.

The file format is as follows:

1.: A file consists of a header and a number of metadata tags divided into sections, each on its own line.
2.: The header is a ;FFMETADATA string, followed by a version number (now 1).
3.: Metadata tags are of the form key=value
4.: Immediately after header follows global metadata
5.: After global metadata there may be sections with per-stream/per-chapter metadata.
6.: A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in brackets ([, ]) and ends with next section or end of file.
7.: At the beginning of a chapter section there may be an optional timebase to be used for start/end values. It must be in form TIMEBASE=num/den, where num and den are integers. If the timebase is missing then start/end times are assumed to be in milliseconds.
Next a chapter section must contain chapter start and end times in form START=num, END=num, where num is a positive integer.
8.: Empty lines and lines starting with ; or # are ignored.
9.: Metadata keys or values containing special characters (=, ;, #, \ and a newline) must be escaped with a backslash \.
10.: Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of the tag (in the example above key is foo , value is
bar).

A ffmetadata file might look like this:

        ;FFMETADATA1
        title=bike\\shed
        ;this is a comment
        artist=FFmpeg troll team
        
        [CHAPTER]
        TIMEBASE=1/1000
        START=0
        #chapter ends at 0:01:00
        END=60000
        title=chapter \#1
        [STREAM]
        title=multi\
        line

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

Extracting an ffmetadata file with ffmpeg goes as follows:

        ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

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

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

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.

The list of supported options follows:

protocol_whitelist list (input): Set a ``,''-separated list of allowed protocols. ``ALL'' matches all protocols. Protocols prefixed by ``-'' are disabled. All protocols are allowed by default but protocols used by an another protocol (nested protocols) are restricted to a per protocol subset.

PROTOCOLS

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

When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list all available ones using the configure option ``--list-protocols''.

You can disable all the protocols using the configure option ``--disable-protocols'', and selectively enable a protocol using the option "--enable-protocol=PROTOCOL``, or you can disable a particular protocol using the option ''--disable-protocol=PROTOCOL".

The option ``-protocols'' of the ff* tools will display the list of supported protocols.

All protocols accept the following options:

rw_timeout: Maximum time to wait for (network) read/write operations to complete, in microseconds.

A description of the currently available protocols follows.

async

Asynchronous data filling wrapper for input stream.

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

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

bluray

Read BluRay playlist.

The accepted options are:

angle: BluRay angle
chapter: Start chapter (1...N)
playlist: Playlist to read (BDMV/PLAYLIST/?????.mpls)

Examples:

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

        bluray:/mnt/bluray

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

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

cache

Caching wrapper for input stream.

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

        cache:<URL>

concat

Physical concatenation protocol.

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

A URL accepted by this protocol has the syntax:

        concat:<URL1>|<URL2>|...|<URLN>

where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one possibly specifying a distinct protocol.

For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with ffplay use the command:

        ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

Note that you may need to escape the character ``|'' which is special for many shells.

crypto

AES-encrypted stream reading protocol.

The accepted options are:

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

Accepted URL formats:

        crypto:<URL>
        crypto+<URL>

data

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

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

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

file

File access protocol.

Read from or write to a file.

A file URL can have the form:

        file:<filename>

where filename is the path of the file to read.

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

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

        ffmpeg -i file:input.mpeg output.mpeg

This protocol accepts the following options:

truncate: Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
blocksize: Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable for files on slow medium.

ftp

FTP (File Transfer Protocol).

Read from or write to remote resources using FTP protocol.

Following syntax is required.

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

This protocol accepts the following options.

timeout: Set timeout in microseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified.
ftp-anonymous-password: Password used when login as anonymous user. Typically an e-mail address should be used.
ftp-write-seekable: Control seekability of connection during encoding. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable. Default value is 0.

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

This protocol accepts the following options:

follow: If set to 1, the protocol will retry reading at the end of the file, allowing reading files that still are being written. In order for this to terminate, you either need to use the rw_timeout option, or use the interrupt callback (for API users).

gopher

Gopher protocol.

hls

Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying "+proto" after the hls URI scheme name, where proto is either ``file'' or ``http''.

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

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

http

HTTP (Hyper Text Transfer Protocol).

This protocol accepts the following options:

seekable

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

chunked_post

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

content_type

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

http_proxy

set HTTP proxy to tunnel through e.g. http://example.com:1234

headers

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

multiple_requests

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

post_data

Set custom HTTP post data.

user_agent

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

user-agent

This is a deprecated option, you can use user_agent instead it.

timeout

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

reconnect_at_eof

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

reconnect_streamed

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

reconnect_delay_max

Sets the maximum delay in seconds after which to give up reconnecting

mime_type

Export the MIME type.

icy

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

icy_metadata_headers

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

icy_metadata_packet

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

cookies

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

offset

Set initial byte offset.

end_offset

Try to limit the request to bytes preceding this offset.

method

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

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

listen

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

        # Server side (sending):
        ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>
        
        # Client side (receiving):
        ffmpeg -i http://<server>:<port> -c copy somefile.ogg
        
        # Client can also be done with wget:
        wget http://<server>:<port> -O somefile.ogg
        
        # Server side (receiving):
        ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg
        
        # Client side (sending):
        ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>
        
        # Client can also be done with wget:
        wget --post-file=somefile.ogg http://<server>:<port>

HTTP Cookies

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

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

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

Icecast

Icecast protocol (stream to Icecast servers)

This protocol accepts the following options:

ice_genre: Set the stream genre.
ice_name: Set the stream name.
ice_description: Set the stream description.
ice_url: Set the stream website URL.
ice_public: Set if the stream should be public. The default is 0 (not public).
user_agent: Override the User-Agent header. If not specified a string of the form ``Lavf/<version>'' will be used.
password: Set the Icecast mountpoint password.
content_type: Set the stream content type. This must be set if it is different from audio/mpeg.
legacy_icecast: This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT method but the SOURCE method.

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

mmst

MMS (Microsoft Media Server) protocol over TCP.

mmsh

MMS (Microsoft Media Server) protocol over HTTP.

The required syntax is:

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

md5

MD5 output protocol.

Computes the MD5 hash of the data to be written, and on close writes this to the designated output or stdout if none is specified. It can be used to test muxers without writing an actual file.

Some examples follow.

        # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
        ffmpeg -i input.flv -f avi -y md5:output.avi.md5
        
        # Write the MD5 hash of the encoded AVI file to stdout.
        ffmpeg -i input.flv -f avi -y md5:

Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with the MD5 output protocol.

pipe

UNIX pipe access protocol.

Read and write from UNIX pipes.

The accepted syntax is:

        pipe:[<number>]

number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not specified, by default the stdout file descriptor will be used for writing, stdin for reading.

For example to read from stdin with ffmpeg:

        cat test.wav | ffmpeg -i pipe:0
        # ...this is the same as...
        cat test.wav | ffmpeg -i pipe:

For writing to stdout with ffmpeg:

        ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
        # ...this is the same as...
        ffmpeg -i test.wav -f avi pipe: | cat > test.avi

This protocol accepts the following options:

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

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

rtmp

Real-Time Messaging Protocol.

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

The required syntax is:

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

The accepted parameters are:

username: An optional username (mostly for publishing).
password: An optional password (mostly for publishing).
server: The address of the RTMP server.
port: The number of the TCP port to use (by default is 1935).
app: It is the name of the application to access. It usually corresponds to the path where the application is installed on the RTMP server (e.g. /ondemand/, /flash/live/, etc.). You can override the value parsed from the URI through the "rtmp_app" option, too.
playpath: It is the path or name of the resource to play with reference to the application specified in app, may be prefixed by ``mp4:''. You can override the value parsed from the URI through the "rtmp_playpath" option, too.
listen: Act as a server, listening for an incoming connection.
timeout: Maximum time to wait for the incoming connection. Implies listen.

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

rtmp_app: Name of application to connect on the RTMP server. This option overrides the parameter specified in the URI.
rtmp_buffer: Set the client buffer time in milliseconds. The default is 3000.
rtmp_conn: Extra arbitrary AMF connection parameters, parsed from a string, e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0". Each value is prefixed by a single character denoting the type, B for Boolean, N for number, S for string, O for object, or Z for null, followed by a colon. For Booleans the data must be either 0 or 1 for FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or 1 to end or begin an object, respectively. Data items in subobjects may be named, by prefixing the type with 'N' and specifying the name before the value (i.e. "NB:myFlag:1"). This option may be used multiple times to construct arbitrary AMF sequences.
rtmp_flashver: Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)
rtmp_flush_interval: Number of packets flushed in the same request (RTMPT only). The default is 10.
rtmp_live: Specify that the media is a live stream. No resuming or seeking in live streams is possible. The default value is "any", which means the subscriber first tries to play the live stream specified in the playpath. If a live stream of that name is not found, it plays the recorded stream. The other possible values are "live" and "recorded".
rtmp_pageurl: URL of the web page in which the media was embedded. By default no value will be sent.
rtmp_playpath: Stream identifier to play or to publish. This option overrides the parameter specified in the URI.
rtmp_subscribe: Name of live stream to subscribe to. By default no value will be sent. It is only sent if the option is specified or if rtmp_live is set to live.
rtmp_swfhash: SHA256 hash of the decompressed SWF file (32 bytes).
rtmp_swfsize: Size of the decompressed SWF file, required for SWFVerification.
rtmp_swfurl: URL of the SWF player for the media. By default no value will be sent.
rtmp_swfverify: URL to player swf file, compute hash/size automatically.
rtmp_tcurl: URL of the target stream. Defaults to proto://host[:port]/app.

For example to read with ffplay a multimedia resource named ``sample'' from the application ``vod'' from an RTMP server ``myserver'':

        ffplay rtmp://myserver/vod/sample

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

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

rtmpe

Encrypted Real-Time Messaging Protocol.

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

rtmps

Real-Time Messaging Protocol over a secure SSL connection.

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

rtmpt

Real-Time Messaging Protocol tunneled through HTTP.

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

rtmpte

Encrypted Real-Time Messaging Protocol tunneled through HTTP.

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

rtmpts

Real-Time Messaging Protocol tunneled through HTTPS.

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

libsmbclient

libsmbclient permits one to manipulate CIFS/SMB network resources.

Following syntax is required.

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

This protocol accepts the following options.

timeout: Set timeout in milliseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified.
truncate: Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
workgroup: Set the workgroup used for making connections. By default workgroup is not specified.

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

libssh

Secure File Transfer Protocol via libssh

Read from or write to remote resources using SFTP protocol.

Following syntax is required.

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

This protocol accepts the following options.

timeout: Set timeout of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified.
truncate: Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
private_key: Specify the path of the file containing private key to use during authorization. By default libssh searches for keys in the ~/.ssh/ directory.

Example: Play a file stored on remote server.

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

librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte

Real-Time Messaging Protocol and its variants supported through librtmp.

Requires the presence of the librtmp headers and library during configuration. You need to explicitly configure the build with ``--enable-librtmp''. If enabled this will replace the native RTMP protocol.

This protocol provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS).

The required syntax is:

        <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>

where rtmp_proto is one of the strings ``rtmp'', ``rtmpt'', ``rtmpe'', ``rtmps'', ``rtmpte'', ``rtmpts'' corresponding to each RTMP variant, and server, port, app and playpath have the same meaning as specified for the RTMP native protocol. options contains a list of space-separated options of the form key=val.

See the librtmp manual page (man 3 librtmp) for more information.

For example, to stream a file in real-time to an RTMP server using ffmpeg:

        ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream

To play the same stream using ffplay:

        ffplay "rtmp://myserver/live/mystream live=1"

rtp

Real-time Transport Protocol.

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

port specifies the RTP port to use.

The following URL options are supported:

ttl=n: Set the TTL (Time-To-Live) value (for multicast only).
rtcpport=n: Set the remote RTCP port to n.
localrtpport=n: Set the local RTP port to n.
localrtcpport=n': Set the local RTCP port to n.
pkt_size=n: Set max packet size (in bytes) to n.
connect=0|1: Do a "connect()" on the UDP socket (if set to 1) or not (if set to 0).
sources=ip[,ip]: List allowed source IP addresses.
block=ip[,ip]: List disallowed (blocked) source IP addresses.
write_to_source=0|1: Send packets to the source address of the latest received packet (if set to 1) or to a default remote address (if set to 0).
localport=n: Set the local RTP port to n.
This is a deprecated option. Instead, localrtpport should be used.

Important notes:

1.: If rtcpport is not set the RTCP port will be set to the RTP port value plus 1.
2.: If localrtpport (the local RTP port) is not set any available port will be used for the local RTP and RTCP ports.
3.: If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port value plus 1.

rtsp

Real-Time Streaming Protocol.

RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with data transferred over RDT).

The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's <https://github.com/revmischa/rtsp-server>).

The required syntax for a RTSP url is:

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

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

The following options are supported.

initial_pause

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

rtsp_transport

Set RTSP transport protocols.

It accepts the following values:

udp: Use UDP as lower transport protocol.
tcp: Use TCP (interleaving within the RTSP control channel) as lower transport protocol.
udp_multicast: Use UDP multicast as lower transport protocol.
http: Use HTTP tunneling as lower transport protocol, which is useful for passing proxies.

: Multiple lower transport protocols may be specified, in that case they are tried one at a time (if the setup of one fails, the next one is tried). For the muxer, only the tcp and udp options are supported.

rtsp_flags

Set RTSP flags.

The following values are accepted:

filter_src: Accept packets only from negotiated peer address and port.
listen: Act as a server, listening for an incoming connection.
prefer_tcp: Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.

: Default value is none.

allowed_media_types

Set media types to accept from the server.

The following flags are accepted:

video
audio
data

: By default it accepts all media types.

min_port

Set minimum local UDP port. Default value is 5000.

max_port

Set maximum local UDP port. Default value is 65000.

timeout

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

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

reorder_queue_size

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

stimeout

Set socket TCP I/O timeout in microseconds.

user-agent

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

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

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

Examples

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

*

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

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

*

Watch a stream tunneled over HTTP:

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

*

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

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

*

Receive a stream in realtime:

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

sap

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

Muxer

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

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

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

announce_addr=address: Specify the destination IP address for sending the announcements to. If omitted, the announcements are sent to the commonly used SAP announcement multicast address 224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.
announce_port=port: Specify the port to send the announcements on, defaults to 9875 if not specified.
ttl=ttl: Specify the time to live value for the announcements and RTP packets, defaults to 255.
same_port=0|1: If set to 1, send all RTP streams on the same port pair. If zero (the default), all streams are sent on unique ports, with each stream on a port 2 numbers higher than the previous. VLC/Live555 requires this to be set to 1, to be able to receive the stream. The RTP stack in libavformat for receiving requires all streams to be sent on unique ports.

Example command lines follow.

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

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

Similarly, for watching in ffplay:

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

And for watching in ffplay, over IPv6:

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

Demuxer

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

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

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

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

Example command lines follow.

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

        ffplay sap://

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

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

sctp

Stream Control Transmission Protocol.

The accepted URL syntax is:

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

The protocol accepts the following options:

listen: If set to any value, listen for an incoming connection. Outgoing connection is done by default.
max_streams: Set the maximum number of streams. By default no limit is set.

srtp

Secure Real-time Transport Protocol.

The accepted options are:

srtp_in_suite

srtp_out_suite

Select input and output encoding suites.

Supported values:

AES_CM_128_HMAC_SHA1_80
SRTP_AES128_CM_HMAC_SHA1_80
AES_CM_128_HMAC_SHA1_32
SRTP_AES128_CM_HMAC_SHA1_32

srtp_in_params

srtp_out_params

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

subfile

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

Accepted options:

start: Start offset of the extracted segment, in bytes.
end: End offset of the extracted segment, in bytes.

Examples:

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

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

Play an AVI file directly from a TAR archive:

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

tee

Writes the output to multiple protocols. The individual outputs are separated by |

        tee:file://path/to/local/this.avi|file://path/to/local/that.avi

tcp

Transmission Control Protocol.

The required syntax for a TCP url is:

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

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

The list of supported options follows.

listen=1|0: Listen for an incoming connection. Default value is 0.
timeout=microseconds: Set raise error timeout, expressed in microseconds.
This option is only relevant in read mode: if no data arrived in more than this time interval, raise error.
listen_timeout=milliseconds: Set listen timeout, expressed in milliseconds.
recv_buffer_size=bytes: Set receive buffer size, expressed bytes.
send_buffer_size=bytes: Set send buffer size, expressed bytes.

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

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

tls

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

The required syntax for a TLS/SSL url is:

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

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

ca_file, cafile=filename: A file containing certificate authority (CA) root certificates to treat as trusted. If the linked TLS library contains a default this might not need to be specified for verification to work, but not all libraries and setups have defaults built in. The file must be in OpenSSL PEM format.
tls_verify=1|0: If enabled, try to verify the peer that we are communicating with. Note, if using OpenSSL, this currently only makes sure that the peer certificate is signed by one of the root certificates in the CA database, but it does not validate that the certificate actually matches the host name we are trying to connect to. (With GnuTLS, the host name is validated as well.)
This is disabled by default since it requires a CA database to be provided by the caller in many cases.
cert_file, cert=filename: A file containing a certificate to use in the handshake with the peer. (When operating as server, in listen mode, this is more often required by the peer, while client certificates only are mandated in certain setups.)
key_file, key=filename: A file containing the private key for the certificate.
listen=1|0: If enabled, listen for connections on the provided port, and assume the server role in the handshake instead of the client role.

Example command lines:

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

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

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

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

udp

User Datagram Protocol.

The required syntax for an UDP URL is:

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

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

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

The list of supported options follows.

buffer_size=size: Set the UDP maximum socket buffer size in bytes. This is used to set either the receive or send buffer size, depending on what the socket is used for. Default is 64KB. See also fifo_size.
bitrate=bitrate: If set to nonzero, the output will have the specified constant bitrate if the input has enough packets to sustain it.
burst_bits=bits: When using bitrate this specifies the maximum number of bits in packet bursts.
localport=port: Override the local UDP port to bind with.
localaddr=addr: Choose the local IP address. This is useful e.g. if sending multicast and the host has multiple interfaces, where the user can choose which interface to send on by specifying the IP address of that interface.
pkt_size=size: Set the size in bytes of UDP packets.
reuse=1|0: Explicitly allow or disallow reusing UDP sockets.
ttl=ttl: Set the time to live value (for multicast only).
connect=1|0: Initialize the UDP socket with "connect()". In this case, the destination address can't be changed with ff_udp_set_remote_url later. If the destination address isn't known at the start, this option can be specified in ff_udp_set_remote_url, too. This allows finding out the source address for the packets with getsockname, and makes writes return with AVERROR(ECONNREFUSED) if ``destination unreachable'' is received. For receiving, this gives the benefit of only receiving packets from the specified peer address/port.
sources=address[,address]: Only receive packets sent to the multicast group from one of the specified sender IP addresses.
block=address[,address]: Ignore packets sent to the multicast group from the specified sender IP addresses.
fifo_size=units: Set the UDP receiving circular buffer size, expressed as a number of packets with size of 188 bytes. If not specified defaults to 7*4096.
overrun_nonfatal=1|0: Survive in case of UDP receiving circular buffer overrun. Default value is 0.
timeout=microseconds: Set raise error timeout, expressed in microseconds.
This option is only relevant in read mode: if no data arrived in more than this time interval, raise error.
broadcast=1|0: Explicitly allow or disallow UDP broadcasting.
Note that broadcasting may not work properly on networks having a broadcast storm protection.

Examples

*

Use ffmpeg to stream over UDP to a remote endpoint:

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

*

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

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

*

Use ffmpeg to receive over UDP from a remote endpoint:

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

unix

Unix local socket

The required syntax for a Unix socket URL is:

        unix://<filepath>

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

timeout: Timeout in ms.
listen: Create the Unix socket in listening mode.

DEVICE OPTIONS

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

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

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

INPUT DEVICES

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

When you configure your FFmpeg build, all the supported input devices are enabled by default. You can list all available ones using the configure option ``--list-indevs''.

You can disable all the input devices using the configure option ``--disable-indevs'', and selectively enable an input device using the option "--enable-indev=INDEV``, or you can disable a particular input device using the option ''--disable-indev=INDEV".

The option ``-devices'' of the ff* tools will display the list of supported input devices.

A description of the currently available input devices follows.

alsa

ALSA (Advanced Linux Sound Architecture) input device.

To enable this input device during configuration you need libasound installed on your system.

This device allows capturing from an ALSA device. The name of the device to capture has to be an ALSA card identifier.

An ALSA identifier has the syntax:

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

where the DEV and SUBDEV components are optional.

The three arguments (in order: CARD,DEV,SUBDEV) specify card number or identifier, device number and subdevice number (-1 means any).

To see the list of cards currently recognized by your system check the files /proc/asound/cards and /proc/asound/devices.

For example to capture with ffmpeg from an ALSA device with card id 0, you may run the command:

        ffmpeg -f alsa -i hw:0 alsaout.wav

For more information see: <http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html>

Options

sample_rate: Set the sample rate in Hz. Default is 48000.
channels: Set the number of channels. Default is 2.

avfoundation

AVFoundation input device.

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

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

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

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

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

and/or

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

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

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

There are two device name aliases:

default: Select the AVFoundation default device of the corresponding type.
none: Do not record the corresponding media type. This is equivalent to specifying an empty device name or index.

Options

AVFoundation supports the following options:

-list_devices <TRUE|FALSE>: If set to true, a list of all available input devices is given showing all device names and indices.
-video_device_index <INDEX>: Specify the video device by its index. Overrides anything given in the input filename.
-audio_device_index <INDEX>: Specify the audio device by its index. Overrides anything given in the input filename.
-pixel_format <FORMAT>: Request the video device to use a specific pixel format. If the specified format is not supported, a list of available formats is given and the first one in this list is used instead. Available pixel formats are: "monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0, bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10, yuv420p, nv12, yuyv422, gray"
-framerate: Set the grabbing frame rate. Default is "ntsc", corresponding to a frame rate of "30000/1001".
-video_size: Set the video frame size.
-capture_cursor: Capture the mouse pointer. Default is 0.
-capture_mouse_clicks: Capture the screen mouse clicks. Default is 0.

Examples

*

Print the list of AVFoundation supported devices and exit:

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

*

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

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

*

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

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

*

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

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

bktr

BSD video input device.

Options

framerate

Set the frame rate.

video_size

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

standard

Available values are:

pal
ntsc
secam
paln
palm
ntscj

decklink

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

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

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

Options

list_devices: If set to true, print a list of devices and exit. Defaults to false.
list_formats: If set to true, print a list of supported formats and exit. Defaults to false.
bm_v210: If set to 1, video is captured in 10 bit v210 instead of uyvy422. Not all Blackmagic devices support this option.
teletext_lines: If set to nonzero, an additional teletext stream will be captured from the vertical ancillary data. This option is a bitmask of the VBI lines checked, 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. Capturing teletext only works for SD PAL sources in 8 bit mode. To use this option, ffmpeg needs to be compiled with "--enable-libzvbi".
channels: Defines number of audio channels to capture. Must be 2, 8 or 16. Defaults to 2.
duplex_mode: Sets the decklink device duplex mode. Must be unset, half or full. Defaults to unset.
video_input: Sets the video input source. Must be unset, sdi, hdmi, optical_sdi, component, composite or s_video. Defaults to unset.
audio_input: Sets the audio input source. Must be unset, embedded, aes_ebu, analog, analog_xlr, analog_rca or microphone. Defaults to unset.
video_pts: Sets the video packet timestamp source. Must be video, audio, reference or wallclock. Defaults to video.
audio_pts: Sets the audio packet timestamp source. Must be video, audio, reference or wallclock. Defaults to audio.
draw_bars: If set to true, color bars are drawn in the event of a signal loss. Defaults to true.

Examples

*

List input devices:

        ffmpeg -f decklink -list_devices 1 -i dummy

*

List supported formats:

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

*

Capture video clip at 1080i50 (format 11):

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

*

Capture video clip at 1080i50 10 bit:

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

*

Capture video clip at 1080i50 with 16 audio channels:

        ffmpeg -channels 16 -f decklink -i 'UltraStudio Mini Recorder@11' -acodec copy -vcodec copy output.avi

dshow

Windows DirectShow input device.

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

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

The input name should be in the format:

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

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

Options

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

video_size: Set the video size in the captured video.
framerate: Set the frame rate in the captured video.
sample_rate: Set the sample rate (in Hz) of the captured audio.
sample_size: Set the sample size (in bits) of the captured audio.
channels: Set the number of channels in the captured audio.
list_devices: If set to true, print a list of devices and exit.
list_options: If set to true, print a list of selected device's options and exit.
video_device_number: Set video device number for devices with the same name (starts at 0, defaults to 0).
audio_device_number: Set audio device number for devices with the same name (starts at 0, defaults to 0).
pixel_format: Select pixel format to be used by DirectShow. This may only be set when the video codec is not set or set to rawvideo.
audio_buffer_size: Set audio device buffer size in milliseconds (which can directly impact latency, depending on the device). Defaults to using the audio device's default buffer size (typically some multiple of 500ms). Setting this value too low can degrade performance. See also <http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx>
video_pin_name: Select video capture pin to use by name or alternative name.
audio_pin_name: Select audio capture pin to use by name or alternative name.
crossbar_video_input_pin_number: Select video input pin number for crossbar device. This will be routed to the crossbar device's Video Decoder output pin. Note that changing this value can affect future invocations (sets a new default) until system reboot occurs.
crossbar_audio_input_pin_number: Select audio input pin number for crossbar device. This will be routed to the crossbar device's Audio Decoder output pin. Note that changing this value can affect future invocations (sets a new default) until system reboot occurs.
show_video_device_dialog: If set to true, before capture starts, popup a display dialog to the end user, allowing them to change video filter properties and configurations manually. Note that for crossbar devices, adjusting values in this dialog may be needed at times to toggle between PAL (25 fps) and NTSC (29.97) input frame rates, sizes, interlacing, etc. Changing these values can enable different scan rates/frame rates and avoiding green bars at the bottom, flickering scan lines, etc. Note that with some devices, changing these properties can also affect future invocations (sets new defaults) until system reboot occurs.
show_audio_device_dialog: If set to true, before capture starts, popup a display dialog to the end user, allowing them to change audio filter properties and configurations manually.
show_video_crossbar_connection_dialog: If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify crossbar pin routings, when it opens a video device.
show_audio_crossbar_connection_dialog: If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify crossbar pin routings, when it opens an audio device.
show_analog_tv_tuner_dialog: If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify TV channels and frequencies.
show_analog_tv_tuner_audio_dialog: If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify TV audio (like mono vs. stereo, Language A,B or C).
audio_device_load: Load an audio capture filter device from file instead of searching it by name. It may load additional parameters too, if the filter supports the serialization of its properties to. To use this an audio capture source has to be specified, but it can be anything even fake one.
audio_device_save: Save the currently used audio capture filter device and its parameters (if the filter supports it) to a file. If a file with the same name exists it will be overwritten.
video_device_load: Load a video capture filter device from file instead of searching it by name. It may load additional parameters too, if the filter supports the serialization of its properties to. To use this a video capture source has to be specified, but it can be anything even fake one.
video_device_save: Save the currently used video capture filter device and its parameters (if the filter supports it) to a file. If a file with the same name exists it will be overwritten.

Examples

*

Print the list of DirectShow supported devices and exit:

        $ ffmpeg -list_devices true -f dshow -i dummy

*

Open video device Camera:

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

*

Open second video device with name Camera:

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

*

Open video device Camera and audio device Microphone:

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

*

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

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

*

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

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

*

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

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

dv1394

Linux DV 1394 input device.

Options

framerate

Set the frame rate. Default is 25.

standard

Available values are:

pal
ntsc

: Default value is "ntsc".

fbdev

Linux framebuffer input device.

The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on a computer monitor, typically on the console. It is accessed through a file device node, usually /dev/fb0.

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

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

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

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

You can take a single screenshot image with the command:

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

Options

framerate: Set the frame rate. Default is 25.

gdigrab

Win32 GDI-based screen capture device.

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

There are two options for the input filename:

        desktop

        title=<window_title>

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

For example, to grab the entire desktop using ffmpeg:

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

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

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

Grab the contents of the window named ``Calculator''

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

Options

draw_mouse

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

framerate

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

show_region

Show grabbed region on screen.

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

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

For example:

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

video_size

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

offset_x

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

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

offset_y

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

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

iec61883

FireWire DV/HDV input device using libiec61883.

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

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

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

Options

dvtype: Override autodetection of DV/HDV. This should only be used if auto detection does not work, or if usage of a different device type should be prohibited. Treating a DV device as HDV (or vice versa) will not work and result in undefined behavior. The values auto, dv and hdv are supported.
dvbuffer: Set maximum size of buffer for incoming data, in frames. For DV, this is an exact value. For HDV, it is not frame exact, since HDV does not have a fixed frame size.
dvguid: Select the capture device by specifying 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 -hdvbuffer 100000 out.mpg

jack

JACK input device.

To enable this input device during configuration you need libjack installed on your system.

A JACK input device creates one or more JACK writable clients, one for each audio channel, with name client_name:input_N, where client_name is the name provided by the application, and N is a number which identifies the channel. Each writable client will send the acquired data to the FFmpeg input device.

Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients.

To connect or disconnect JACK clients you can use the jack_connect and jack_disconnect programs, or do it through a graphical interface, for example with qjackctl.

To list the JACK clients and their properties you can invoke the command jack_lsp.

Follows an example which shows how to capture a JACK readable client with ffmpeg.

        # Create a JACK writable client with name "ffmpeg".
        $ ffmpeg -f jack -i ffmpeg -y out.wav
        
        # Start the sample jack_metro readable client.
        $ jack_metro -b 120 -d 0.2 -f 4000
        
        # List the current JACK clients.
        $ jack_lsp -c
        system:capture_1
        system:capture_2
        system:playback_1
        system:playback_2
        ffmpeg:input_1
        metro:120_bpm
        
        # Connect metro to the ffmpeg writable client.
        $ jack_connect metro:120_bpm ffmpeg:input_1

For more information read: <http://jackaudio.org/>

Options

channels: Set the number of channels. Default is 2.

lavfi

Libavfilter input virtual device.

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

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

Options

graph

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

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

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

graph_file

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

dumpgraph

Dump graph to stderr.

Examples

*

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

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

*

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

        ffplay -f lavfi color=c=pink

*

Create three different video test filtered sources and play them:

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

*

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

        ffplay -f lavfi "amovie=test.wav"

*

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

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

*

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

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

libcdio

Audio-CD input device based on libcdio.

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

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

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

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

Options

speed

Set drive reading speed. Default value is 0.

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

paranoia_mode

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

disable
verify
overlap
neverskip
full

Default value is disable.

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

libdc1394

IIDC1394 input device, based on libdc1394 and libraw1394.

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

openal

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

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

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

An incomplete list of OpenAL implementations follows:

Creative: The official Windows implementation, providing hardware acceleration with supported devices and software fallback. See <http://openal.org/>.
OpenAL Soft: Portable, open source (LGPL) software implementation. Includes backends for the most common sound APIs on the Windows, Linux, Solaris, and BSD operating systems. See <http://kcat.strangesoft.net/openal.html>.
Apple: OpenAL is part of Core Audio, the official Mac OS X Audio interface. See <http://developer.apple.com/technologies/mac/audio-and-video.html>

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

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

Options

channels: Set the number of channels in the captured audio. Only the values 1 (monaural) and 2 (stereo) are currently supported. Defaults to 2.
sample_size: Set the sample size (in bits) of the captured audio. Only the values 8 and 16 are currently supported. Defaults to 16.
sample_rate: Set the sample rate (in Hz) of the captured audio. Defaults to 44.1k.
list_devices: If set to true, print a list of devices and exit. Defaults to false.

Examples

Print the list of OpenAL supported devices and exit:

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

Capture from the OpenAL device DR-BT101 via PulseAudio:

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

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

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

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

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

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

oss

Open Sound System input device.

The filename to provide to the input device is the device node representing the OSS input device, and is usually set to /dev/dsp.

For example to grab from /dev/dsp using ffmpeg use the command:

        ffmpeg -f oss -i /dev/dsp /tmp/oss.wav

For more information about OSS see: <http://manuals.opensound.com/usersguide/dsp.html>

Options

sample_rate: Set the sample rate in Hz. Default is 48000.
channels: Set the number of channels. Default is 2.

pulse

PulseAudio input device.

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

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

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

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

Options

server: Connect to a specific PulseAudio server, specified by an IP address. Default server is used when not provided.
name: Specify the application name PulseAudio will use when showing active clients, by default it is the "LIBAVFORMAT_IDENT" string.
stream_name: Specify the stream name PulseAudio will use when showing active streams, by default it is ``record''.
sample_rate: Specify the samplerate in Hz, by default 48kHz is used.
channels: Specify the channels in use, by default 2 (stereo) is set.
frame_size: Specify the number of bytes per frame, by default it is set to 1024.
fragment_size: Specify the minimal buffering fragment in PulseAudio, it will affect the audio latency. By default it is unset.
wallclock: Set the initial PTS using the current time. Default is 1.

Examples

Record a stream from default device:

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

qtkit

QTKit input device.

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

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


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


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


        
        ffmpeg -f qtkit -list_devices true -i ""

Options

frame_rate: Set frame rate. Default is 30.
list_devices: If set to "true", print a list of devices and exit. Default is "false".
video_device_index: Select the video device by index for devices with the same name (starts at 0).

sndio

sndio input device.

To enable this input device during configuration you need libsndio installed on your system.

The filename to provide to the input device is the device node representing the sndio input device, and is usually set to /dev/audio0.

For example to grab from /dev/audio0 using ffmpeg use the command:

        ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav

Options

sample_rate: Set the sample rate in Hz. Default is 48000.
channels: Set the number of channels. Default is 2.

video4linux2, v4l2

Video4Linux2 input video device.

``v4l2'' can be used as alias for ``video4linux2''.

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

The name of the device to grab is a file device node, usually Linux systems tend to automatically create such nodes when the device (e.g. an USB webcam) is plugged into the system, and has a name of the kind /dev/videoN, where N is a number associated to the device.

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

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

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

*

List supported formats for a video4linux2 device:

        ffplay -f video4linux2 -list_formats all /dev/video0

*

Grab and show the input of a video4linux2 device:

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

*

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

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

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

Options

standard

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

channel

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

video_size

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

pixel_format

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

input_format

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

framerate

Set the preferred video frame rate.

list_formats

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

Available values are:

all: Show all available (compressed and non-compressed) formats.
raw: Show only raw video (non-compressed) formats.
compressed: Show only compressed formats.

list_standards

List supported standards and exit.

Available values are:

all: Show all supported standards.

timestamps, ts

Set type of timestamps for grabbed frames.

Available values are:

default: Use timestamps from the kernel.
abs: Use absolute timestamps (wall clock).
mono2abs: Force conversion from monotonic to absolute timestamps.

: Default value is "default".

use_libv4l2

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

vfwcap

VfW (Video for Windows) capture input device.

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

Options

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

x11grab

X11 video input device.

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

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

This device allows one to capture a region of an X11 display.

The filename passed as input has the syntax:

        [<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]

hostname:display_number.screen_number specifies the X11 display name of the screen to grab from. hostname can be omitted, and defaults to ``localhost''. The environment variable DISPLAY contains the default display name.

x_offset and y_offset specify the offsets of the grabbed area with respect to the top-left border of the X11 screen. They default to 0.

Check the X11 documentation (e.g. man X) for more detailed information.

Use the xdpyinfo program for getting basic information about the properties of your X11 display (e.g. grep for ``name'' or ``dimensions'').

For example to grab from :0.0 using ffmpeg:

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

Grab at position "10,20":

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

Options

draw_mouse

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

follow_mouse

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

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

For example:

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

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

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

framerate

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

show_region

Show grabbed region on screen.

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

region_border

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

For example:

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

With follow_mouse:

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

video_size

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

use_shm

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

grab_x

grab_y

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

caca

CACA output device.

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

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

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

Options

window_title: Set the CACA window title, if not specified default to the filename specified for the output device.
window_size: Set the CACA window size, can be a string of the form widthxheight or a video size abbreviation. If not specified it defaults to the size of the input video.
driver: Set display driver.
algorithm: Set dithering algorithm. Dithering is necessary because the picture being rendered has usually far more colours than the available palette. The accepted values are listed with "-list_dither algorithms".
antialias: Set antialias method. Antialiasing smoothens the rendered image and avoids the commonly seen staircase effect. The accepted values are listed with "-list_dither antialiases".
charset: Set which characters are going to be used when rendering text. The accepted values are listed with "-list_dither charsets".
color: Set color to be used when rendering text. The accepted values are listed with "-list_dither colors".
list_drivers: If set to true, print a list of available drivers and exit.
list_dither: List available dither options related to the argument. The argument must be one of "algorithms", "antialiases", "charsets", "colors".

Examples

*

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

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

*

Show the list of available drivers and exit:

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

*

Show the list of available dither colors and exit:

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

decklink

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

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

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

Options

list_devices: If set to true, print a list of devices and exit. Defaults to false.
list_formats: If set to true, print a list of supported formats and exit. Defaults to false.
preroll: Amount of time to preroll video in seconds. Defaults to 0.5.

Examples

*

List output devices:

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

*

List supported formats:

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

*

Play video clip:

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

*

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

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

fbdev

Linux framebuffer output device.

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

Options

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

Examples

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

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

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

opengl

OpenGL output device.

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

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

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

Options

background: Set background color. Black is a default.
no_window: Disables default SDL window when set to non-zero value. Application must provide OpenGL context and both "window_size_cb" and "window_swap_buffers_cb" callbacks when set.
window_title: Set the SDL window title, if not specified default to the filename specified for the output device. Ignored when no_window is set.
window_size: Set preferred window size, can be a string of the form widthxheight or a video size abbreviation. If not specified it defaults to the size of the input video, downscaled according to the aspect ratio. Mostly usable when no_window is not set.

Examples

Play a file on SDL window using OpenGL rendering:

        ffmpeg  -i INPUT -f opengl "window title"

oss

OSS (Open Sound System) output device.

pulse

PulseAudio output device.

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

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

Options

server

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

name

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

stream_name

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

device

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

buffer_size

buffer_duration

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

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

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

prebuf

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

minreq

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

Examples

Play a file on default device on default server:

        ffmpeg  -i INPUT -f pulse "stream name"

sdl

SDL (Simple DirectMedia Layer) output device.

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

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

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

Options

window_title: Set the SDL window title, if not specified default to the filename specified for the output device.
icon_title: Set the name of the iconified SDL window, if not specified it is set to the same value of window_title.
window_size: Set the SDL window size, can be a string of the form widthxheight or a video size abbreviation. If not specified it defaults to the size of the input video, downscaled according to the aspect ratio.
window_fullscreen: Set fullscreen mode when non-zero value is provided. Default value is zero.

Interactive commands

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

q, ESC: Quit the device immediately.

Examples

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

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

sndio

sndio audio output device.

xv

XV (XVideo) output device.

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

Options

display_name

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

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

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

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

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

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

window_id

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

window_size

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

window_x

window_y

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

window_title

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

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

Examples

*

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

        ffmpeg -i INPUT OUTPUT -f xv display

*

Decode and display the input video to multiple X11 windows:

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

RESAMPLER OPTIONS

The audio resampler supports the following named options.

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

ich, in_channel_count

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

och, out_channel_count

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

uch, used_channel_count

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

isr, in_sample_rate

Set the input sample rate. Default value is 0.

osr, out_sample_rate

Set the output sample rate. Default value is 0.

isf, in_sample_fmt

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

osf, out_sample_fmt

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

tsf, internal_sample_fmt

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

icl, in_channel_layout

ocl, out_channel_layout

Set the input/output channel layout.

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

clev, center_mix_level

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

slev, surround_mix_level

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

lfe_mix_level

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

rmvol, rematrix_volume

Set rematrix volume. Default value is 1.0.

rematrix_maxval

Set maximum output value for rematrixing. This can be used to prevent clipping vs. preventing 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 if set to 1, default value is 0.

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 disabled.

cutoff

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

precision

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

cheby

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

async

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

first_pts

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

min_comp

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

min_hard_comp

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

comp_duration

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

max_soft_comp

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

matrix_encoding

Select matrixed stereo encoding.

It accepts the following values:

none: select none
dolby: select Dolby
dplii: select Dolby Pro Logic II

: Default value is "none".

filter_type

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

It accepts the following values:

cubic: select cubic
blackman_nuttall: select Blackman Nuttall windowed sinc
kaiser: select Kaiser windowed sinc

kaiser_beta

For swr only, set Kaiser window beta value. Must be a double float value in the interval [2,16], default value is 9.

output_sample_bits

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

SCALER OPTIONS

The video scaler supports the following named options.

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

sws_flags

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

It accepts the following values:

fast_bilinear: Select fast bilinear scaling algorithm.
bilinear: Select bilinear scaling algorithm.
bicubic: Select bicubic scaling algorithm.
experimental: Select experimental scaling algorithm.
neighbor: Select nearest neighbor rescaling algorithm.
area: Select averaging area rescaling algorithm.
bicublin: Select bicubic scaling algorithm for the luma component, bilinear for chroma components.
gauss: Select Gaussian rescaling algorithm.
sinc: Select sinc rescaling algorithm.
lanczos: Select Lanczos rescaling algorithm.
spline: Select natural bicubic spline rescaling algorithm.
print_info: Enable printing/debug logging.
accurate_rnd: Enable accurate rounding.
full_chroma_int: Enable full chroma interpolation.
full_chroma_inp: Select full chroma input.
bitexact: Enable bitexact output.

srcw

Set source width.

srch

Set source height.

dstw

Set destination width.

dsth

Set destination height.

src_format

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

dst_format

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

src_range

Select source range.

dst_range

Select destination range.

param0, param1

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

sws_dither

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

auto: automatic choice
none: no dithering
bayer: bayer dither
ed: error diffusion dither
a_dither: arithmetic dither, based using addition
x_dither: arithmetic dither, based using xor (more random/less apparent patterning that a_dither).

alphablend

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

uniform_color: Blend onto a uniform background color
checkerboard: Blend onto a checkerboard
none: No blending

FILTERING INTRODUCTION

Filtering in FFmpeg is enabled through the libavfilter library.

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

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

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

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

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

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

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

Some filters take in input a list of parameters: they are specified after the filter name and an equal sign, and are separated from each other by a colon.

There exist so-called source filters that do not have an audio/video input, and sink filters that will not have audio/video output.

GRAPH

The graph2dot program included in the FFmpeg tools directory can be used to parse a filtergraph description and issue a corresponding textual representation in the dot language.

Invoke the command:

        graph2dot -h

to see how to use graph2dot.

You can then pass the dot description to the dot program (from the graphviz suite of programs) and obtain a graphical representation of the filtergraph.

For example the sequence of commands:

        echo <GRAPH_DESCRIPTION> | \
        tools/graph2dot -o graph.tmp && \
        dot -Tpng graph.tmp -o graph.png && \
        display graph.png

can be used to create and display an image representing the graph described by the GRAPH_DESCRIPTION string. Note that this string must be a complete self-contained graph, with its inputs and outputs explicitly defined. For example if your command line is of the form:

        ffmpeg -i infile -vf scale=640:360 outfile

your GRAPH_DESCRIPTION string will need to be of the form:

        nullsrc,scale=640:360,nullsink

you may also need to set the nullsrc parameters and add a format filter in order to simulate a specific input file.

FILTERGRAPH DESCRIPTION

A filtergraph is a directed graph of connected filters. It can contain cycles, and there can be multiple links between a pair of filters. Each link has one input pad on one side connecting it to one filter from which it takes its input, and one output pad on the other side connecting it to one filter accepting its output.

Each filter in a filtergraph is an instance of a filter class registered in the application, which defines the features and the number of input and output pads of the filter.

A filter with no input pads is called a ``source'', and a filter with no output pads is called a ``sink''.

Filtergraph syntax

A filtergraph has a textual representation, which is recognized by the -filter/-vf/-af and -filter_complex options in ffmpeg and -vf/-af in ffplay, and by the "avfilter_graph_parse_ptr()" function defined in libavfilter/avfilter.h.

A filterchain consists of a sequence of connected filters, each one connected to the previous one in the sequence. A filterchain is represented by a list of ``,''-separated filter descriptions.

A filtergraph consists of a sequence of filterchains. A sequence of filterchains is represented by a list of ``;''-separated filterchain descriptions.

A filter is represented by a string of the form: [in_link_1]...[in_link_N]filter_name=arguments[out_link_1]...[out_link_M]

filter_name is the name of the filter class of which the described filter is an instance of, and has to be the name of one of the filter classes registered in the program. The name of the filter class is optionally followed by a string "=arguments".

arguments is a string which contains the parameters used to initialize the filter instance. It may have one of two forms:

*: A ':'-separated list of key=value pairs.
*: A ':'-separated list of value. In this case, the keys are assumed to be the option names in the order they are declared. E.g. the "fade" filter declares three options in this order --- type, start_frame and nb_frames. Then the parameter list in:0:30 means that the value in is assigned to the option type, 0 to start_frame and 30 to nb_frames.
*: A ':'-separated list of mixed direct value and long key=value pairs. The direct value must precede the key=value pairs, and follow the same constraints order of the previous point. The following key=value pairs can be set in any preferred order.

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

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

The name and arguments of the filter are optionally preceded and followed by a list of link labels. A link label allows one to name a link and associate it to a filter output or input pad. The preceding labels in_link_1 ... in_link_N, are associated to the filter input pads, the following labels out_link_1 ... out_link_M, are associated to the output pads.

When two link labels with the same name are found in the filtergraph, a link between the corresponding input and output pad is created.

If an output pad is not labelled, it is linked by default to the first unlabelled input pad of the next filter in the filterchain. For example in the filterchain

        nullsrc, split[L1], [L2]overlay, nullsink

the split filter instance has two output pads, and the overlay filter instance two input pads. The first output pad of split is labelled ``L1'', the first input pad of overlay is labelled ``L2'', and the second output pad of split is linked to the second input pad of overlay, which are both unlabelled.

In a filter description, if the input label of the first filter is not specified, ``in'' is assumed; if the output label of the last filter is not specified, ``out'' is assumed.

In a complete filterchain all the unlabelled filter input and output pads must be connected. A filtergraph is considered valid if all the filter input and output pads of all the filterchains are connected.

Libavfilter will automatically insert scale filters where format conversion is required. It is possible to specify swscale flags for those automatically inserted scalers by prepending "sws_flags=flags;" to the filtergraph description.

Here is a BNF description of the filtergraph syntax:

        <NAME>             ::= sequence of alphanumeric characters and '_'
        <LINKLABEL>        ::= "[" <NAME> "]"
        <LINKLABELS>       ::= <LINKLABEL> [<LINKLABELS>]
        <FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
        <FILTER>           ::= [<LINKLABELS>] <NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
        <FILTERCHAIN>      ::= <FILTER> [,<FILTERCHAIN>]
        <FILTERGRAPH>      ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

Notes on filtergraph escaping

Filtergraph description composition entails several levels of escaping. See the ``Quoting and escaping'' section in the ffmpeg-utils(1) manual for more information about the employed escaping procedure.

A first level escaping affects the content of each filter option value, which may contain the special character ":" used to separate values, or one of the escaping characters "\'".

A second level escaping affects the whole filter description, which may contain the escaping characters "\'" or the special characters "[],;" used by the filtergraph description.

Finally, when you specify a filtergraph on a shell commandline, you need to perform a third level escaping for the shell special characters contained within it.

For example, consider the following string to be embedded in the drawtext filter description text value:

        this is a 'string': may contain one, or more, special characters

This string contains the "'" special escaping character, and the ":" special character, so it needs to be escaped in this way:

        text=this is a \'string\'\: may contain one, or more, special characters

A second level of escaping is required when embedding the filter description in a filtergraph description, in order to escape all the filtergraph special characters. Thus the example above becomes:

        drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters

(note that in addition to the "\'" escaping special characters, also "," needs to be escaped).

Finally an additional level of escaping is needed when writing the filtergraph description in a shell command, which depends on the escaping rules of the adopted shell. For example, assuming that "\" is special and needs to be escaped with another "\", the previous string will finally result in:

        -vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"

TIMELINE EDITING

Some filters support a generic enable option. For the filters supporting timeline editing, this option can be set to an expression which is evaluated before sending a frame to the filter. If the evaluation is non-zero, the filter will be enabled, otherwise the frame will be sent unchanged to the next filter in the filtergraph.

The expression accepts the following values:

t: timestamp expressed in seconds, NAN if the input timestamp is unknown
n: sequential number of the input frame, starting from 0
pos: the position in the file of the input frame, NAN if unknown
w
h: width and height of the input frame if video

Additionally, these filters support an enable command that can be used to re-define the expression.

Like any other filtering option, the enable option follows the same rules.

For example, to enable a blur filter (smartblur) from 10 seconds to 3 minutes, and a curves filter starting at 3 seconds:

        smartblur = enable='between(t,10,3*60)',
        curves    = enable='gte(t,3)' : preset=cross_process

AUDIO FILTERS

When you configure your FFmpeg build, you can disable any of the existing filters using "--disable-filters". The configure output will show the audio filters included in your build.

Below is a description of the currently available audio filters.

acompressor

A compressor is mainly used to reduce the dynamic range of a signal. Especially modern music is mostly compressed at a high ratio to improve the overall loudness. It's done to get the highest attention of a listener, ``fatten'' the sound and bring more ``power'' to the track. If a signal is compressed too much it may sound dull or ``dead'' afterwards or it may start to ``pump'' (which could be a powerful effect but can also destroy a track completely). The right compression is the key to reach a professional sound and is the high art of mixing and mastering. Because of its complex settings it may take a long time to get the right feeling for this kind of effect.

Compression is done by detecting the volume above a chosen level "threshold" and dividing it by the factor set with "ratio". So if you set the threshold to -12dB and your signal reaches -6dB a ratio of 2:1 will result in a signal at -9dB. Because an exact manipulation of the signal would cause distortion of the waveform the reduction can be levelled over the time. This is done by setting ``Attack'' and ``Release''. "attack" determines how long the signal has to rise above the threshold before any reduction will occur and "release" sets the time the signal has to fall below the threshold to reduce the reduction again. Shorter signals than the chosen attack time will be left untouched. The overall reduction of the signal can be made up afterwards with the "makeup" setting. So compressing the peaks of a signal about 6dB and raising the makeup to this level results in a signal twice as loud than the source. To gain a softer entry in the compression the "knee" flattens the hard edge at the threshold in the range of the chosen decibels.

The filter accepts the following options:

level_in: Set input gain. Default is 1. Range is between 0.015625 and 64.
threshold: If a signal of second stream rises above this level it will affect the gain reduction of the first stream. 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 2. Range is from 1 and 64.
knee: Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.
link: Choose if the "average" level between all channels of 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.

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

acrusher

Reduce audio bit resolution.

This filter is bit crusher with enhanced functionality. A bit crusher is used to audibly reduce number of bits an audio signal is sampled with. This doesn't change the bit depth at all, it just produces the effect. Material reduced in bit depth sounds more harsh and ``digital''. This filter is able to even round to continuous values instead of discrete bit depths. Additionally it has a D/C offset which results in different crushing of the lower and the upper half of the signal. An Anti-Aliasing setting is able to produce ``softer'' crushing sounds.

Another feature of this filter is the logarithmic mode. This setting switches from linear distances between bits to logarithmic ones. The result is a much more ``natural'' sounding crusher which doesn't gate low signals for example. The human ear has a logarithmic perception, too so this kind of crushing is much more pleasant. Logarithmic crushing is also able to get anti-aliased.

The filter accepts the following options:

level_in: Set level in.
level_out: Set level out.
bits: Set bit reduction.
mix: Set mixing amount.
mode: Can be linear: "lin" or logarithmic: "log".
dc: Set DC.
aa: Set anti-aliasing.
samples: Set sample reduction.
lfo: Enable LFO. By default disabled.
lforange: Set LFO range.
lforate: Set LFO rate.

adelay

Delay one or more audio channels.

Samples in delayed channel are filled with silence.

The filter accepts the following option:

delays: Set list of delays in milliseconds for each channel separated by '|'. At least one delay greater than 0 should be provided. Unused delays will be silently ignored. If number of given delays is smaller than number of channels all remaining channels will not be delayed. If you want to delay exact number of samples, append 'S' to number.

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

aecho

Apply echoing to the input audio.

Echoes are reflected sound and can occur naturally amongst mountains (and sometimes large buildings) when talking or shouting; digital echo effects emulate this behaviour and are often used to help fill out the sound of a single instrument or vocal. The time difference between the original signal and the reflection is the "delay", and the loudness of the reflected signal is the "decay". Multiple echoes can have different delays and decays.

A description of the accepted parameters follows.

in_gain: Set input gain of reflected signal. Default is 0.6.
out_gain: Set output gain of reflected signal. Default is 0.3.
delays: Set list of time intervals in milliseconds between original signal and reflections separated by '|'. Allowed range for each "delay" is "(0 - 90000.0]". Default is 1000.
decays: Set list of loudnesses of reflected signals separated by '|'. Allowed range for each "decay" is "(0 - 1.0]". Default is 0.5.

Examples

*

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

        aecho=0.8:0.88:60:0.4

*

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

        aecho=0.8:0.88:6:0.4

*

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

        aecho=0.8:0.9:1000:0.3

*

Same as above but with one more mountain:

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

aemphasis

Audio emphasis filter creates or restores material directly taken from LPs or emphased CDs with different filter curves. E.g. to store music on vinyl the signal has to be altered by a filter first to even out the disadvantages of this recording medium. Once the material is played back the inverse filter has to be applied to restore the distortion of the frequency response.

The filter accepts the following options:

level_in

Set input gain.

level_out

Set output gain.

mode

Set filter mode. For restoring material use "reproduction" mode, otherwise use "production" mode. Default is "reproduction" mode.

type

Set filter type. Selects medium. Can be one of the following:

col: select Columbia.
emi: select EMI.
bsi: select BSI (78RPM).
riaa: select RIAA.
cd: select Compact Disc (CD).
50fm: select 50Xs (FM).
75fm: select 75Xs (FM).
50kf: select 50Xs (FM-KF).
75kf: select 75Xs (FM-KF).

aeval

Modify an audio signal according to the specified expressions.

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

It accepts the following parameters:

exprs: Set the '|'-separated expressions list for each separate channel. If the number of input channels is greater than the number of expressions, the last specified expression is used for the remaining output channels.
channel_layout, c: Set output channel layout. If not specified, the channel layout is specified by the number of expressions. If set to same, it will use by default the same input channel layout.

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

ch: channel number of the current expression
n: number of the evaluated sample, starting from 0
s: sample rate
t: time of the evaluated sample expressed in seconds
nb_in_channels
nb_out_channels: input and output number of channels
val(CH): the value of input channel with number CH

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

Examples

*

Half volume:

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

*

Invert phase of the second channel:

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

afade

Apply fade-in/out effect to input audio.

A description of the accepted parameters follows.

type, t

Specify the effect type, can be either "in" for fade-in, or "out" for a fade-out effect. Default is "in".

start_sample, ss

Specify the number of the start sample for starting to apply the fade effect. Default is 0.

nb_samples, ns

Specify the number of samples for which the fade effect has to last. At the end of the fade-in effect the output audio will have the same volume as the input audio, at the end of the fade-out transition the output audio will be silence. Default is 44100.

start_time, st

Specify the start time of the fade effect. Default is 0. The value must be specified as a time duration; see the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. If set this option is used instead of start_sample.

duration, d

Specify the duration of the fade effect. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. At the end of the fade-in effect the output audio will have the same volume as the input audio, at the end of the fade-out transition the output audio will be silence. By default the duration is determined by nb_samples. If set this option is used instead of nb_samples.

curve

Set curve for fade transition.

It accepts the following values:

tri: select triangular, linear slope (default)
qsin: select quarter of sine wave
hsin: select half of sine wave
esin: select exponential sine wave
log: select logarithmic
ipar: select inverted parabola
qua: select quadratic
cub: select cubic
squ: select square root
cbr: select cubic root
par: select parabola
exp: select exponential
iqsin: select inverted quarter of sine wave
ihsin: select inverted half of sine wave
dese: select double-exponential seat
desi: select double-exponential sigmoid

Examples

*

Fade in first 15 seconds of audio:

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

*

Fade out last 25 seconds of a 900 seconds audio:

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

afftfilt

Apply arbitrary expressions to samples in frequency domain.

real

Set frequency domain real expression for each separate channel separated by '|'. Default is ``1''. If the number of input channels is greater than the number of expressions, the last specified expression is used for the remaining output channels.

imag

Set frequency domain imaginary expression for each separate channel separated by '|'. If not set, real option is used.

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

sr: sample rate
b: current frequency bin number
nb: number of available bins
ch: channel number of the current expression
chs: number of channels
pts: current frame pts

win_size

Set window size.

It accepts the following values:

w16
w32
w64
w128
w256
w512
w1024
w2048
w4096
w8192
w16384
w32768
w65536

: Default is "w4096"

win_func

Set window function. Default is "hann".

overlap

Set window overlap. If set to 1, the recommended overlap for selected window function will be picked. Default is 0.75.

Examples

*

Leave almost only low frequencies in audio:

        afftfilt="1-clip((b/nb)*b,0,1)"

aformat

Set output format constraints for the input audio. The framework will negotiate the most appropriate format to minimize conversions.

It accepts the following parameters:

sample_fmts: A '|'-separated list of requested sample formats.
sample_rates: A '|'-separated list of requested sample rates.
channel_layouts: A '|'-separated list of requested channel layouts.
See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

If a parameter is omitted, all values are allowed.

Force the output to either unsigned 8-bit or signed 16-bit stereo

        aformat=sample_fmts=u8|s16:channel_layouts=stereo

agate

A gate is mainly used to reduce lower parts of a signal. This kind of signal processing reduces disturbing noise between useful signals.

Gating is done by detecting the volume below a chosen level threshold and dividing it by the factor set with ratio. The bottom of the noise floor is set via range. Because an exact manipulation of the signal would cause distortion of the waveform the reduction can be levelled over time. This is done by setting attack and release.

attack determines how long the signal has to fall below the threshold before any reduction will occur and release sets the time the signal has to rise above the threshold to reduce the reduction again. Shorter signals than the chosen attack time will be left untouched.

level_in: Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
range: Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1.
threshold: If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.
ratio: Set a ratio by which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
attack: Amount of milliseconds the signal has to rise above the threshold before gain reduction stops. Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
release: Amount of milliseconds the signal has to fall below the threshold before the reduction is increased again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.
makeup: Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
knee: Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.
detection: Choose if exact signal should be taken for detection or an RMS like one. Default is "rms". Can be "peak" or "rms".
link: Choose if the average level between all channels or the louder channel affects the reduction. Default is "average". Can be "average" or "maximum".

alimiter

The limiter prevents an input signal from rising over a desired threshold. This limiter uses lookahead technology to prevent your signal from distorting. It means that there is a small delay after the signal is processed. Keep in mind that the delay it produces is the attack time you set.

The filter accepts the following options:

level_in: Set input gain. Default is 1.
level_out: Set output gain. Default is 1.
limit: Don't let signals above this level pass the limiter. Default is 1.
attack: The limiter will reach its attenuation level in this amount of time in milliseconds. Default is 5 milliseconds.
release: Come back from limiting to attenuation 1.0 in this amount of milliseconds. Default is 50 milliseconds.
asc: When gain reduction is always needed ASC takes care of releasing to an average reduction level rather than reaching a reduction of 0 in the release time.
asc_level: Select how much the release time is affected by ASC, 0 means nearly no changes in release time while 1 produces higher release times.
level: Auto level output signal. Default is enabled. This normalizes audio back to 0dB if enabled.

Depending on picked setting it is recommended to upsample input 2x or 4x times with aresample before applying this filter.

allpass

Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-width width. An all-pass filter changes the audio's frequency to phase relationship without changing its frequency to amplitude relationship.

The filter accepts the following options:

frequency, f

Set frequency in Hz.

width_type

Set method to specify band-width of filter.

h: Hz
q: Q-Factor
o: octave
s: slope

width, w

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

aloop

Loop audio samples.

The filter accepts the following options:

loop: Set the number of loops.
size: Set maximal number of samples.
start: Set first sample of loop.

amerge

Merge two or more audio streams into a single multi-channel stream.

The filter accepts the following options:

inputs: Set the number of inputs. Default is 2.

If the channel layouts of the inputs are disjoint, and therefore compatible, the channel layout of the output will be set accordingly and the channels will be reordered as necessary. If the channel layouts of the inputs are not disjoint, the output will have all the channels of the first input then all the channels of the second input, in that order, and the channel layout of the output will be the default value corresponding to the total number of channels.

For example, if the first input is in 2.1 (FL+FR+LF) and the second input is FC+BL+BR, then the output will be in 5.1, with the channels in the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the first input, b1 is the first channel of the second input).

On the other hand, if both input are in stereo, the output channels will be in the default order: a1, a2, b1, b2, and the channel layout will be arbitrarily set to 4.0, which may or may not be the expected value.

All inputs must have the same sample rate, and format.

If inputs do not have the same duration, the output will stop with the shortest.

Examples

*

Merge two mono files into a stereo stream:

        amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge

*

Multiple merges assuming 1 video stream and 6 audio streams in input.mkv:

        ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv

amix

Mixes multiple audio inputs into a single output.

Note that this filter only supports float samples (the amerge and pan audio filters support many formats). If the amix input has integer samples then aresample will be automatically inserted to perform the conversion to float samples.

For example

        ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT

will mix 3 input audio streams to a single output with the same duration as the first input and a dropout transition time of 3 seconds.

It accepts the following parameters:

inputs

The number of inputs. If unspecified, it defaults to 2.

duration

How to determine the end-of-stream.

longest: The duration of the longest input. (default)
shortest: The duration of the shortest input.
first: The duration of the first input.

dropout_transition

The transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.

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 herz. gain set new gain parameter in dB.

Full filter invocation with asendcmd may look like this: asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...

anull

Pass the audio source unchanged to the output.

apad

Pad the end of an audio stream with silence.

This can be used together with ffmpeg -shortest to extend audio streams to the same length as the video stream.

A description of the accepted options follows.

packet_size: Set silence packet size. Default value is 4096.
pad_len: Set the number of samples of silence to add to the end. After the value is reached, the stream is terminated. This option is mutually exclusive with whole_len.
whole_len: Set the minimum total number of samples in the output audio stream. If the value is longer than the input audio length, silence is added to the end, until the value is reached. This option is mutually exclusive with pad_len.

If neither the pad_len nor the whole_len option is set, the filter will add silence to the end of the input stream indefinitely.

Examples

*

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

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 ffmpeg-resampler manual for the complete list of supported options.

Examples

*

Resample the input audio to 44100Hz:

        aresample=44100

*

Stretch/squeeze samples to the given timestamps, with a maximum of 1000 samples per second compensation:

        aresample=async=1000

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

asetnsamples

Set the number of samples per each output audio frame.

The last output packet may contain a different number of samples, as the filter will flush all the remaining samples when the input audio signals its end.

The filter accepts the following options:

nb_out_samples, n: Set the number of frames per each output audio frame. The number is intended as the number of samples per each channel. Default value is 1024.
pad, p: If set to 1, the filter will pad the last audio frame with zeroes, so that the last frame will contain the same number of samples as the previous ones. Default value is 1.

For example, to set the number of per-frame samples to 1234 and disable padding for the last frame, use:

        asetnsamples=n=1234:p=0

asetrate

Set the sample rate without altering the PCM data. This will result in a change of speed and pitch.

The filter accepts the following options:

sample_rate, r: Set the output sample rate. Default is 44100 Hz.

ashowinfo

Show a line containing various information for each input audio frame. The input audio is not modified.

The shown line contains a sequence of key/value pairs of the form key:value.

The following values are shown in the output:

n: The (sequential) number of the input frame, starting from 0.
pts: The presentation timestamp of the input frame, in time base units; the time base depends on the filter input pad, and is usually 1/sample_rate.
pts_time: The presentation timestamp of the input frame in seconds.
pos: position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic audio)
fmt: The sample format.
chlayout: The channel layout.
rate: The sample rate for the audio frame.
nb_samples: The number of samples (per channel) in the frame.
checksum: The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio, the data is treated as if all the planes were concatenated.
plane_checksums: A list of Adler-32 checksums for each data plane.

astats

Display time domain statistical information about the audio channels. Statistics are calculated and displayed for each audio channel and, where applicable, an overall figure is also given.

It accepts the following option:

length

Short window length in seconds, used for peak and trough RMS measurement. Default is 0.05 (50 milliseconds). Allowed range is "[0.1 - 10]".

metadata

Set metadata injection. All the metadata keys are prefixed with "lavfi.astats.X", where "X" is channel number starting from 1 or string "Overall". Default is disabled.

Available keys for each channel are: DC_offset Min_level Max_level Min_difference Max_difference Mean_difference Peak_level RMS_peak RMS_trough Crest_factor Flat_factor Peak_count Bit_depth

and for Overall: DC_offset Min_level Max_level Min_difference Max_difference Mean_difference Peak_level RMS_level RMS_peak RMS_trough Flat_factor Peak_count Bit_depth Number_of_samples

For example full key look like this "lavfi.astats.1.DC_offset" or this "lavfi.astats.Overall.Peak_count".

For description what each key means read below.

reset

Set number of frame after which stats are going to be recalculated. Default is disabled.

A description of each shown parameter follows:

DC offset: Mean amplitude displacement from zero.
Min level: Minimal sample level.
Max level: Maximal sample level.
Min difference: Minimal difference between two consecutive samples.
Max difference: Maximal difference between two consecutive samples.
Mean difference: Mean difference between two consecutive samples. The average of each difference between two consecutive samples.
Peak level dB
RMS level dB: Standard peak and RMS level measured in dBFS.
RMS peak dB
RMS trough dB: Peak and trough values for RMS level measured over a short window.
Crest factor: Standard ratio of peak to RMS level (note: not in dB).
Flat factor: Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels (i.e. either Min level or Max level).
Peak count: Number of occasions (not the number of samples) that the signal attained either Min level or Max level.
Bit depth: Overall bit depth of audio. Number of bits used for each sample.

asyncts

Synchronize audio data with timestamps by squeezing/stretching it and/or dropping samples/adding silence when needed.

This filter is not built by default, please use aresample to do squeezing/stretching.

It accepts the following parameters:

compensate: Enable stretching/squeezing the data to make it match the timestamps. Disabled by default. When disabled, time gaps are covered with silence.
min_delta: The minimum difference between timestamps and audio data (in seconds) to trigger adding/dropping samples. The default value is 0.1. If you get an imperfect sync with this filter, try setting this parameter to 0.
max_comp: The maximum compensation in samples per second. Only relevant with compensate=1. The default value is 500.
first_pts: Assume that the first PTS should be this value. The time base is 1 / sample rate. This allows for padding/trimming at the start of the stream. By default, no assumption is made about the first frame's expected PTS, so no padding or trimming is done. For example, this could be set to 0 to pad the beginning with silence if an audio stream starts after the video stream or to trim any samples with a negative PTS due to encoder delay.

atempo

Adjust audio tempo.

The filter accepts exactly one parameter, the audio tempo. If not specified then the filter will assume nominal 1.0 tempo. Tempo must be in the [0.5, 2.0] range.

Examples

*

Slow down audio to 80% tempo:

        atempo=0.8

*

To speed up audio to 125% tempo:

        atempo=1.25

atrim

Trim the input so that the output contains one continuous subpart of the input.

It accepts the following parameters:

start: Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with the timestamp start will be the first sample in the output.
end: Specify time of the first audio sample that will be dropped, i.e. the audio sample immediately preceding the one with the timestamp end will be the last sample in the output.
start_pts: Same as start, except this option sets the start timestamp in samples instead of seconds.
end_pts: Same as end, except this option sets the end timestamp in samples instead of seconds.
duration: The maximum duration of the output in seconds.
start_sample: The number of the first sample that should be output.
end_sample: The number of the first sample that should be dropped.

start, end, and duration are expressed as time duration specifications; see the Time duration section in the ffmpeg-utils(1) manual.

Note that the first two sets of the start/end options and the duration option look at the frame timestamp, while the _sample options simply count the samples that pass through the filter. So start/end_pts and start/end_sample will give different results when the timestamps are wrong, inexact or do not start at zero. Also note that this filter does not modify the timestamps. If you wish to have the output timestamps start at zero, insert the asetpts filter after the atrim filter.

If multiple start or end options are set, this filter tries to be greedy and keep all samples that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple atrim filters.

The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time.

Examples:

*

Drop everything except the second minute of input:

        ffmpeg -i INPUT -af atrim=60:120

*

Keep only the first 1000 samples:

        ffmpeg -i INPUT -af atrim=end_sample=1000

bandpass

Apply a two-pole Butterworth band-pass filter with central frequency frequency, and (3dB-point) band-width width. The csg option selects a constant skirt gain (peak gain = Q) instead of the default: constant 0dB peak gain. The filter roll off at 6dB per octave (20dB per decade).

The filter accepts the following options:

frequency, f

Set the filter's central frequency. Default is 3000.

csg

Constant skirt gain if set to 1. Defaults to 0.

width_type

Set method to specify band-width of filter.

h: Hz
q: Q-Factor
o: octave
s: slope

width, w

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

bandreject

Apply a two-pole Butterworth band-reject filter with central frequency frequency, and (3dB-point) band-width width. The filter roll off at 6dB per octave (20dB per decade).

The filter accepts the following options:

frequency, f

Set the filter's central frequency. Default is 3000.

width_type

Set method to specify band-width of filter.

h: Hz
q: Q-Factor
o: octave
s: slope

width, w

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

bass

Boost or cut the bass (lower) frequencies of the audio using a two-pole shelving filter with a response similar to that of a standard hi-fi's tone-controls. This is also known as shelving equalisation (EQ).

The filter accepts the following options:

gain, g

Give the gain at 0 Hz. Its useful range is about -20 (for a large cut) to +20 (for a large boost). Beware of clipping when using a positive gain.

frequency, f

Set the filter's central frequency and so can be used to extend or reduce the frequency range to be boosted or cut. The default value is 100 Hz.

width_type

Set method to specify band-width of filter.

h: Hz
q: Q-Factor
o: octave
s: slope

width, w

Determine how steep is the filter's shelf transition.

biquad

Apply a biquad IIR filter with the given coefficients. Where b0, b1, b2 and a0, a1, a2 are the numerator and denominator coefficients respectively.

bs2b

Bauer stereo to binaural transformation, which improves headphone listening of stereo audio records.

It accepts the following parameters:

profile

Pre-defined crossfeed level.

default: Default level (fcut=700, feed=50).
cmoy: Chu Moy circuit (fcut=700, feed=60).
jmeier: Jan Meier circuit (fcut=650, feed=95).

fcut

Cut frequency (in Hz).

feed

Feed level (in Hz).

channelmap

Remap input channels to new locations.

It accepts the following parameters:

channel_layout: The channel layout of the output stream.
map: Map channels from input to output. The argument is a '|'-separated list of mappings, each in the "in_channel-out_channel" or in_channel form. in_channel can be either the name of the input channel (e.g. FL for front left) or its index in the input channel layout. out_channel is the name of the output channel or its index in the output channel layout. If out_channel is not given then it is implicitly an index, starting with zero and increasing by one for each mapping.

If no mapping is present, the filter will implicitly map input channels to output channels, preserving indices.

For example, assuming a 5.1+downmix input MOV file,

        ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav

will create an output WAV file tagged as stereo from the downmix channels of the input.

To fix a 5.1 WAV improperly encoded in AAC's native channel order

        ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav

channelsplit

Split each channel from an input audio stream into a separate output stream.

It accepts the following parameters:

channel_layout: The channel layout of the input stream. The default is ``stereo''.

For example, assuming a stereo input MP3 file,

        ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv

will create an output Matroska file with two audio streams, one containing only the left channel and the other the right channel.

Split a 5.1 WAV file into per-channel files:

        ffmpeg -i in.wav -filter_complex
        'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
        -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
        front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
        side_right.wav

chorus

Add a chorus effect to the audio.

Can make a single vocal sound like a chorus, but can also be applied to instrumentation.

Chorus resembles an echo effect with a short delay, but whereas with echo the delay is constant, with chorus, it is varied using using sinusoidal or triangular modulation. The modulation depth defines the range the modulated delay is played before or after the delay. Hence the delayed sound will sound slower or faster, that is the delayed sound tuned around the original one, like in a chorus where some vocals are slightly off key.

It accepts the following parameters:

in_gain: Set input gain. Default is 0.4.
out_gain: Set output gain. Default is 0.4.
delays: Set delays. A typical delay is around 40ms to 60ms.
decays: Set decays.
speeds: Set speeds.
depths: Set depths.

Examples

*

A single delay:

        chorus=0.7:0.9:55:0.4:0.25:2

*

Two delays:

        chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3

*

Fuller sounding chorus with three delays:

        chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3

compand

Compress or expand the audio's dynamic range.

It accepts the following parameters:

attacks
decays: A list of times in seconds for each channel over which the instantaneous level of the input signal is averaged to determine its volume. attacks refers to increase of volume and decays refers to decrease of volume. For most situations, the attack time (response to the audio getting louder) should be shorter than the decay time, because the human ear is more sensitive to sudden loud audio than sudden soft audio. A typical value for attack is 0.3 seconds and a typical value for decay is 0.8 seconds. If specified number of attacks & decays is lower than number of channels, the last set attack/decay will be used for all remaining channels.
points: A list of points for the transfer function, specified in dB relative to the maximum possible signal amplitude. Each key points list must be defined using the following syntax: "x0/y0|x1/y1|x2/y2|...." or "x0/y0 x1/y1 x2/y2 ...."
The input values must be in strictly increasing order but the transfer function does not have to be monotonically rising. The point "0/0" is assumed but may be overridden (by "0/out-dBn"). Typical values for the transfer function are "-70/-70|-60/-20".
soft-knee: Set the curve radius in dB for all joints. It defaults to 0.01.
gain: Set the additional gain in dB to be applied at all points on the transfer function. This allows for easy adjustment of the overall gain. It defaults to 0.
volume: Set an initial volume, in dB, to be assumed for each channel when filtering starts. This permits the user to supply a nominal level initially, so that, for example, a very large gain is not applied to initial signal levels before the companding has begun to operate. A typical value for audio which is initially quiet is -90 dB. It defaults to 0.
delay: Set a delay, in seconds. The input audio is analyzed immediately, but audio is delayed before being fed to the volume adjuster. Specifying a delay approximately equal to the attack/decay times allows the filter to effectively operate in predictive rather than reactive mode. It defaults to 0.

Examples

*

Make music with both quiet and loud passages suitable for listening to in a noisy environment:

        compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2

Another example for audio with whisper and explosion parts:

        compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0

*

A noise gate for when the noise is at a lower level than the signal:

        compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1

*

Here is another noise gate, this time for when the noise is at a higher level than the signal (making it, in some ways, similar to squelch):

        compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1

*

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 location. Because the front of sound wave has fixed speed in normal conditions, the phasing of microphones can vary and depends on their location and interposition. The best sound mix can be achieved when these microphones are in phase (synchronized). Note that distance of ~30 cm between microphones makes one microphone to capture signal in antiphase to another microphone. That makes the final mix sounding moody. This filter helps to solve phasing problems by adding different delays to each microphone track and make them synchronized.

The best result can be reached when you take one track as base and synchronize other tracks one by one with it. Remember that synchronization/delay tolerance depends on sample rate, too. Higher sample rates will give more tolerance.

It accepts the following parameters:

mm: Set millimeters distance. This is compensation distance for fine tuning. Default is 0.
cm: Set cm distance. This is compensation distance for tightening distance setup. Default is 0.
m: Set meters distance. This is compensation distance for hard distance setup. Default is 0.
dry: Set dry amount. Amount of unprocessed (dry) signal. Default is 0.
wet: Set wet amount. Amount of processed (wet) signal. Default is 1.
temp: Set temperature degree in Celsius. This is the temperature of the environment. Default is 20.

crystalizer

Simple algorithm to expand audio dynamic range.

The filter accepts the following options:

i: Sets the intensity of effect (default: 2.0). Must be in range between 0.0 (unchanged sound) to 10.0 (maximum effect).
c: Enable clipping. By default is enabled.

dcshift

Apply a DC shift to the audio.

This can be useful to remove a DC offset (caused perhaps by a hardware problem in the recording chain) from the audio. The effect of a DC offset is reduced headroom and hence volume. The astats filter can be used to determine if a signal has a DC offset.

shift: Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift the audio.
limitergain: Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to prevent clipping.

dynaudnorm

Dynamic Audio Normalizer.

This filter applies a certain amount of gain to the input audio in order to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in contrast to more ``simple'' normalization algorithms, the Dynamic Audio Normalizer *dynamically* re-adjusts the gain factor to the input audio. This allows for applying extra gain to the ``quiet'' sections of the audio while avoiding distortions or clipping the ``loud'' sections. In other words: The Dynamic Audio Normalizer will ``even out'' the volume of quiet and loud sections, in the sense that the volume of each section is brought to the same target level. Note, however, that the Dynamic Audio Normalizer achieves this goal *without* applying ``dynamic range compressing''. It will retain 100% of the dynamic range *within* each section of the audio file.

f: Set the frame length in milliseconds. In range from 10 to 8000 milliseconds. Default is 500 milliseconds. The Dynamic Audio Normalizer processes the input audio in small chunks, referred to as frames. This is required, because a peak magnitude has no meaning for just a single sample value. Instead, we need to determine the peak magnitude for a contiguous sequence of sample values. While a ``standard'' normalizer would simply use the peak magnitude of the complete file, the Dynamic Audio Normalizer determines the peak magnitude individually for each frame. The length of a frame is specified in milliseconds. By default, the Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has been found to give good results with most files. Note that the exact frame length, in number of samples, will be determined automatically, based on the sampling rate of the individual input audio file.
g: Set the Gaussian filter window size. In range from 3 to 301, must be odd number. Default is 31. Probably the most important parameter of the Dynamic Audio Normalizer is the "window size" of the Gaussian smoothing filter. The filter's window size is specified in frames, centered around the current frame. For the sake of simplicity, this must be an odd number. Consequently, the default value of 31 takes into account the current frame, as well as the 15 preceding frames and the 15 subsequent frames. Using a larger window results in a stronger smoothing effect and thus in less gain variation, i.e. slower gain adaptation. Conversely, using a smaller window results in a weaker smoothing effect and thus in more gain variation, i.e. faster gain adaptation. In other words, the more you increase this value, the more the Dynamic Audio Normalizer will behave like a ``traditional'' normalization filter. On the contrary, the more you decrease this value, the more the Dynamic Audio Normalizer will behave like a dynamic range compressor.
p: Set the target peak value. This specifies the highest permissible magnitude level for the normalized audio input. This filter will try to approach the target peak magnitude as closely as possible, but at the same time it also makes sure that the normalized signal will never exceed the peak magnitude. A frame's maximum local gain factor is imposed directly by the target peak magnitude. The default value is 0.95 and thus leaves a headroom of 5%*. It is not recommended to go above this value.
m: Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0. The Dynamic Audio Normalizer determines the maximum possible (local) gain factor for each input frame, i.e. the maximum gain factor that does not result in clipping or distortion. The maximum gain factor is determined by the frame's highest magnitude sample. However, the Dynamic Audio Normalizer additionally bounds the frame's maximum gain factor by a predetermined (global) maximum gain factor. This is done in order to avoid excessive gain factors in ``silent'' or almost silent frames. By default, the maximum gain factor is 10.0, For most inputs the default value should be sufficient and it usually is not recommended to increase this value. Though, for input with an extremely low overall volume level, it may be necessary to allow even higher gain factors. Note, however, that the Dynamic Audio Normalizer does not simply apply a ``hard'' threshold (i.e. cut off values above the threshold). Instead, a ``sigmoid'' threshold function will be applied. This way, the gain factors will smoothly approach the threshold value, but never exceed that value.
r: Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled. By default, the Dynamic Audio Normalizer performs ``peak'' normalization. This means that the maximum local gain factor for each frame is defined (only) by the frame's highest magnitude sample. This way, the samples can be amplified as much as possible without exceeding the maximum signal level, i.e. without clipping. Optionally, however, the Dynamic Audio Normalizer can also take into account the frame's root mean square, abbreviated RMS. In electrical engineering, the RMS is commonly used to determine the power of a time-varying signal. It is therefore considered that the RMS is a better approximation of the ``perceived loudness'' than just looking at the signal's peak magnitude. Consequently, by adjusting all frames to a constant RMS value, a uniform ``perceived loudness'' can be established. If a target RMS value has been specified, a frame's local gain factor is defined as the factor that would result in exactly that RMS value. Note, however, that the maximum local gain factor is still restricted by the frame's highest magnitude sample, in order to prevent clipping.
n: Enable channels coupling. By default is enabled. By default, the Dynamic Audio Normalizer will amplify all channels by the same amount. This means the same gain factor will be applied to all channels, i.e. the maximum possible gain factor is determined by the ``loudest'' channel. However, in some recordings, it may happen that the volume of the different channels is uneven, e.g. one channel may be ``quieter'' than the other one(s). In this case, this option can be used to disable the channel coupling. This way, the gain factor will be determined independently for each channel, depending only on the individual channel's highest magnitude sample. This allows for harmonizing the volume of the different channels.
c: Enable DC bias correction. By default is disabled. An audio signal (in the time domain) is a sequence of sample values. In the Dynamic Audio Normalizer these sample values are represented in the -1.0 to 1.0 range, regardless of the original input format. Normally, the audio signal, or ``waveform'', should be centered around the zero point. That means if we calculate the mean value of all samples in a file, or in a single frame, then the result should be 0.0 or at least very close to that value. If, however, there is a significant deviation of the mean value from 0.0, in either positive or negative direction, this is referred to as a DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic Audio Normalizer provides optional DC bias correction. With DC bias correction enabled, the Dynamic Audio Normalizer will determine the mean value, or ``DC correction'' offset, of each input frame and subtract that value from all of the frame's sample values which ensures those samples are centered around 0.0 again. Also, in order to avoid ``gaps'' at the frame boundaries, the DC correction offset values will be interpolated smoothly between neighbouring frames.
b: Enable alternative boundary mode. By default is disabled. The Dynamic Audio Normalizer takes into account a certain neighbourhood around each frame. This includes the preceding frames as well as the subsequent frames. However, for the ``boundary'' frames, located at the very beginning and at the very end of the audio file, not all neighbouring frames are available. In particular, for the first few frames in the audio file, the preceding frames are not known. And, similarly, for the last few frames in the audio file, the subsequent frames are not known. Thus, the question arises which gain factors should be assumed for the missing frames in the ``boundary'' region. The Dynamic Audio Normalizer implements two modes to deal with this situation. The default boundary mode assumes a gain factor of exactly 1.0 for the missing frames, resulting in a smooth ``fade in'' and ``fade out'' at the beginning and at the end of the input, respectively.
s: Set the compress factor. In range from 0.0 to 30.0. Default is 0.0. By default, the Dynamic Audio Normalizer does not apply ``traditional'' compression. This means that signal peaks will not be pruned and thus the full dynamic range will be retained within each local neighbourhood. However, in some cases it may be desirable to combine the Dynamic Audio Normalizer's normalization algorithm with a more ``traditional'' compression. For this purpose, the Dynamic Audio Normalizer provides an optional compression (thresholding) function. If (and only if) the compression feature is enabled, all input frames will be processed by a soft knee thresholding function prior to the actual normalization process. Put simply, the thresholding function is going to prune all samples whose magnitude exceeds a certain threshold value. However, the Dynamic Audio Normalizer does not simply apply a fixed threshold value. Instead, the threshold value will be adjusted for each individual frame. In general, smaller parameters result in stronger compression, and vice versa. Values below 3.0 are not recommended, because audible distortion may appear.

earwax

Make audio easier to listen to on headphones.

This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio so that when listened to on headphones the stereo image is moved from inside your head (standard for headphones) to outside and in front of the listener (standard for speakers).

Ported from SoX.

equalizer

Apply a two-pole peaking equalisation (EQ) filter. With this filter, the signal-level at and around a selected frequency can be increased or decreased, whilst (unlike bandpass and bandreject filters) that at all other frequencies is unchanged.

In order to produce complex equalisation curves, this filter can be given several times, each with a different central frequency.

The filter accepts the following options:

frequency, f

Set the filter's central frequency in Hz.

width_type

Set method to specify band-width of filter.

h: Hz
q: Q-Factor
o: octave
s: slope

width, w

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

gain, g

Set the required gain or attenuation in dB. Beware of clipping when using a positive gain.

Examples

*

Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:

        equalizer=f=1000:width_type=h:width=200:g=-10

*

Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:

        equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g=-5

extrastereo

Linearly increases the difference between left and right channels which adds some sort of ``live'' effect to playback.

The filter accepts the following options:

m: Sets the difference coefficient (default: 2.5). 0.0 means mono sound (average of both channels), with 1.0 sound will be unchanged, with -1.0 left and right channels will be swapped.
c: Enable clipping. By default is enabled.

firequalizer

Apply FIR Equalization using arbitrary frequency response.

The filter accepts the following option:

gain

Set gain curve equation (in dB). The expression can contain variables:

f: the evaluated frequency
sr: sample rate
ch: channel number, set to 0 when multichannels evaluation is disabled
chid: channel id, see libavutil/channel_layout.h, set to the first channel id when multichannels evaluation is disabled
chs: number of channels
chlayout: channel_layout, see libavutil/channel_layout.h

and functions:

gain_interpolate(f): interpolate gain on frequency f based on gain_entry
cubic_interpolate(f): same as gain_interpolate, but smoother

: This option is also available as command. Default is gain_interpolate(f).

gain_entry

Set gain entry for gain_interpolate function. The expression can contain functions:

entry(f, g): store gain entry at frequency f with value g

: This option is also available as command.

delay

Set filter delay in seconds. Higher value means more accurate. Default is 0.01.

accuracy

Set filter accuracy in Hz. Lower value means more accurate. Default is 5.

wfunc

Set window function. Acceptable values are:

rectangular: rectangular window, useful when gain curve is already smooth
hann: hann window (default)
hamming: hamming window
blackman: blackman window
nuttall3: 3-terms continuous 1st derivative nuttall window
mnuttall3: minimum 3-terms discontinuous nuttall window
nuttall: 4-terms continuous 1st derivative nuttall window
bnuttall: minimum 4-terms discontinuous nuttall (blackman-nuttall) window
bharris: blackman-harris window
tukey: tukey window

fixed

If enabled, use fixed number of audio samples. This improves speed when filtering with large delay. Default is disabled.

multi

Enable multichannels evaluation on gain. Default is disabled.

zero_phase

Enable zero phase mode by subtracting timestamp to compensate delay. Default is disabled.

scale

Set scale used by gain. Acceptable values are:

linlin: linear frequency, linear gain
linlog: linear frequency, logarithmic (in dB) gain (default)
loglin: logarithmic (in octave scale where 20 Hz is 0) frequency, linear gain
loglog: logarithmic frequency, logarithmic gain

dumpfile

Set file for dumping, suitable for gnuplot.

dumpscale

Set scale for dumpfile. Acceptable values are same with scale option. Default is linlog.

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 swep delay in milliseconds. Range from 0 to 10. Default value is 2.
regen: Set percentage regeneration (delayed signal feedback). Range from -95 to 95. Default value is 0.
width: Set percentage of delayed signal mixed with original. Range from 0 to 100. Default value is 71.
speed: Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
shape: Set swept wave shape, can be triangular or sinusoidal. Default value is sinusoidal.
phase: Set swept wave percentage-shift for multi channel. Range from 0 to 100. Default value is 25.
interp: Set delay-line interpolation, linear or quadratic. Default is linear.

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 -acodec 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

highpass

Apply a high-pass filter with 3dB point frequency. The filter can be either single-pole, or double-pole (the default). The filter roll off at 6dB per pole per octave (20dB per pole per decade).

The filter accepts the following options:

frequency, f

Set frequency in Hz. Default is 3000.

poles, p

Set number of poles. Default is 2.

width_type

Set method to specify band-width of filter.

h: Hz
q: Q-Factor
o: octave
s: slope

width, w

Specify the band-width of a filter in width_type units. Applies only to double-pole filter. The default is 0.707q and gives a Butterworth response.

join

Join multiple input streams into one multi-channel stream.

It accepts the following parameters:

inputs: The number of input streams. It defaults to 2.
channel_layout: The desired output channel layout. It defaults to stereo.
map: Map channels from inputs to output. The argument is a '|'-separated list of mappings, each in the "input_idx.in_channel-out_channel" form. input_idx is the 0-based index of the input stream. in_channel can be either the name of the input channel (e.g. FL for front left) or its index in the specified input stream. out_channel is the name of the output channel.

The filter will attempt to guess the mappings when they are not specified explicitly. It does so by first trying to find an unused matching input channel and if that fails it picks the first unused input channel.

Join 3 inputs (with properly set channel layouts):

        ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT

Build a 5.1 output from 6 single-channel streams:

        ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
        'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
        out

ladspa

Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.

To enable compilation of this filter you need to configure FFmpeg with "--enable-ladspa".

file, f: Specifies the name of LADSPA plugin library to load. If the environment variable LADSPA_PATH is defined, the LADSPA plugin is searched in each one of the directories specified by the colon separated list in LADSPA_PATH, otherwise in the standard LADSPA paths, which are in this order: HOME/.ladspa/lib/, /usr/local/lib/ladspa/, /usr/lib/ladspa/.
plugin, p: Specifies the plugin within the library. Some libraries contain only one plugin, but others contain many of them. If this is not set filter will list all available plugins within the specified library.
controls, c: Set the '|' separated list of controls which are zero or more floating point values that determine the behavior of the loaded plugin (for example delay, threshold or gain). Controls need to be defined using the following syntax: c0=value0|c1=value1|c2=value2|..., where valuei is the value set on the i-th control. Alternatively they can be also defined using the following syntax: value0|value1|value2|..., where valuei is the value set on the i-th control. If controls is set to "help", all available controls and their valid ranges are printed.
sample_rate, s: Specify the sample rate, default to 44100. Only used if plugin have zero inputs.
nb_samples, n: Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.
duration, d: Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be greater than the specified duration, as the generated audio is always cut at the end of a complete frame. If not specified, or the expressed duration is negative, the audio is supposed to be generated forever. Only used if plugin have zero inputs.

Examples

*

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

Commands

This filter supports the following commands:

cN: Modify the N-th control value.
If the specified value is not valid, it is ignored and prior one is kept.

loudnorm

EBU R128 loudness normalization. Includes both dynamic and linear normalization modes. Support for both single pass (livestreams, files) and double pass (files) modes. This algorithm can target IL, LRA, and maximum true peak.

To enable compilation of this filter you need to configure FFmpeg with "--enable-libebur128".

The filter accepts the following options:

I, i: Set integrated loudness target. Range is -70.0 - -5.0. Default value is -24.0.
LRA, lra: Set loudness range target. Range is 1.0 - 20.0. Default value is 7.0.
TP, tp: Set maximum true peak. Range is -9.0 - +0.0. Default value is -2.0.
measured_I, measured_i: Measured IL of input file. Range is -99.0 - +0.0.
measured_LRA, measured_lra: Measured LRA of input file. Range is 0.0 - 99.0.
measured_TP, measured_tp: Measured true peak of input file. Range is -99.0 - +99.0.
measured_thresh: Measured threshold of input file. Range is -99.0 - +0.0.
offset: Set offset gain. Gain is applied before the true-peak limiter. Range is -99.0 - +99.0. Default is +0.0.
linear: Normalize linearly if possible. measured_I, measured_LRA, measured_TP, and measured_thresh must also to be specified in order to use this mode. Options are true or false. Default is true.
dual_mono: Treat mono input files as ``dual-mono''. If a mono file is intended for playback on a stereo system, its EBU R128 measurement will be perceptually incorrect. If set to "true", this option will compensate for this effect. Multi-channel input files are not affected by this option. Options are true or false. Default is false.
print_format: Set print format for stats. Options are summary, json, or none. Default value is none.

lowpass

Apply a low-pass filter with 3dB point frequency. The filter can be either single-pole or double-pole (the default). The filter roll off at 6dB per pole per octave (20dB per pole per decade).

The filter accepts the following options:

frequency, f

Set frequency in Hz. Default is 500.

poles, p

Set number of poles. Default is 2.

width_type

Set method to specify band-width of filter.

h: Hz
q: Q-Factor
o: octave
s: slope

width, w

Specify the band-width of a filter in width_type units. Applies only to double-pole filter. The default is 0.707q and gives a Butterworth response.

pan

Mix channels with specific gain levels. The filter accepts the output channel layout followed by a set of channels definitions.

This filter is also designed to efficiently remap the channels of an audio stream.

The filter accepts parameters of the form: "l|outdef|outdef|..."

l: output channel layout or number of channels
outdef: output channel specification, of the form: "out_name=[gain*]in_name[+[gain*]in_name...]"
out_name: output channel to define, either a channel name (FL, FR, etc.) or a channel number (c0, c1, etc.)
gain: multiplicative coefficient for the channel, 1 leaving the volume unchanged
in_name: input channel to use, see out_name for details; it is not possible to mix named and numbered input channels

If the `=' in a channel specification is replaced by `<', then the gains for that specification will be renormalized so that the total is 1, thus avoiding clipping noise.

Mixing examples

For example, if you want to down-mix from stereo to mono, but with a bigger factor for the left channel:

        pan=1c|c0=0.9*c0+0.1*c1

A customized down-mix to stereo that works automatically for 3-, 4-, 5- and 7-channels surround:

        pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR

Note that ffmpeg integrates a default down-mix (and up-mix) system that should be preferred (see ``-ac'' option) unless you have very specific needs.

Remapping examples

The channel remapping will be effective if, and only if:

*<gain coefficients are zeroes or ones,>
*<only one input per channel output,>

If all these conditions are satisfied, the filter will notify the user (``Pure channel mapping detected''), and use an optimized and lossless method to do the remapping.

For example, if you have a 5.1 source and want a stereo audio stream by dropping the extra channels:

        pan="stereo| c0=FL | c1=FR"

Given the same source, you can also switch front left and front right channels and keep the input channel layout:

        pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"

If the input is a stereo audio stream, you can mute the front left channel (and still keep the stereo channel layout) with:

        pan="stereo|c1=c1"

Still with a stereo audio stream input, you can copy the right channel in both front left and right:

        pan="stereo| c0=FR | c1=FR"

replaygain

ReplayGain scanner filter. This filter takes an audio stream as an input and outputs it unchanged. At end of filtering it displays "track_gain" and "track_peak".

resample

Convert the audio sample format, sample rate and channel layout. It is not meant to be used directly.

rubberband

Apply time-stretching and pitch-shifting with librubberband.

The filter accepts the following options:

tempo

Set tempo scale factor.

pitch

Set pitch scale factor.

transients

Set transients detector. Possible values are:

crisp
mixed
smooth

detector

Set detector. Possible values are:

compound
percussive
soft

phase

Set phase. Possible values are:

laminar
independent

window

Set processing window size. Possible values are:

standard
short
long

smoothing

Set smoothing. Possible values are:

off
on

formant

Enable formant preservation when shift pitching. Possible values are:

shifted
preserved

pitchq

Set pitch quality. Possible values are:

quality
speed
consistency

channels

Set channels. Possible values are:

apart
together

sidechaincompress

This filter acts like normal compressor but has the ability to compress detected signal using second input signal. It needs two input streams and returns one output stream. First input stream will be processed depending on second stream signal. The filtered signal then can be filtered with other filters in later stages of processing. See pan and amerge filter.

The filter accepts the following options:

level_in: Set input gain. Default is 1. Range is between 0.015625 and 64.
threshold: If a signal of second stream raises above this level it will affect the gain reduction of first stream. By default is 0.125. Range is between 0.00097563 and 1.
ratio: Set a ratio about which the signal is reduced. 1:2 means that if the level raised 4dB above the threshold, it will be only 2dB above after the reduction. Default is 2. Range is between 1 and 20.
attack: Amount of milliseconds the signal has to rise above the threshold before gain reduction starts. Default is 20. Range is between 0.01 and 2000.
release: Amount of milliseconds the signal has to fall below the threshold before reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
makeup: Set the amount by how much signal will be amplified after processing. Default is 2. Range is from 1 and 64.
knee: Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.
link: Choose if the "average" level between all channels of side-chain stream or the louder("maximum") channel of side-chain stream affects the reduction. Default is "average".
detection: Should the exact signal be taken in case of "peak" or an RMS one in case of "rms". Default is "rms" which is mainly smoother.
level_sc: Set sidechain gain. Default is 1. Range is between 0.015625 and 64.
mix: How much to use compressed signal in output. Default is 1. Range is between 0 and 1.

Examples

*

Full ffmpeg example taking 2 audio inputs, 1st input to be compressed depending on the signal of 2nd input and later compressed signal to be merged with 2nd input:

        ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"

sidechaingate

A sidechain gate acts like a normal (wideband) gate but has the ability to filter the detected signal before sending it to the gain reduction stage. Normally a gate uses the full range signal to detect a level above the threshold. For example: If you cut all lower frequencies from your sidechain signal the gate will decrease the volume of your track only if not enough highs appear. With this technique you are able to reduce the resonation of a natural drum or remove ``rumbling'' of muted strokes from a heavily distorted guitar. It needs two input streams and returns one output stream. First input stream will be processed depending on second stream signal.

The filter accepts the following options:

level_in: Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
range: Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1.
threshold: If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.
ratio: Set a ratio about which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
attack: Amount of milliseconds the signal has to rise above the threshold before gain reduction stops. Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
release: Amount of milliseconds the signal has to fall below the threshold before the reduction is increased again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.
makeup: Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
knee: Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.
detection: Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak or rms.
link: Choose if the average level between all channels or the louder channel affects the reduction. Default is average. Can be average or maximum.
level_sc: Set sidechain gain. Default is 1. Range is from 0.015625 to 64.

silencedetect

Detect silence in an audio stream.

This filter logs a message when it detects that the input audio volume is less or equal to a noise tolerance value for a duration greater or equal to the minimum detected noise duration.

The printed times and duration are expressed in seconds.

The filter accepts the following options:

duration, d: Set silence duration until notification (default is 2 seconds).
noise, n: Set noise tolerance. Can be specified in dB (in case ``dB'' is appended to the specified value) or amplitude ratio. Default is -60dB, or 0.001.

Examples

*

Detect 5 seconds of silence with -50dB noise tolerance:

        silencedetect=n=-50dB:d=5

*

Complete example with ffmpeg to detect silence with 0.0001 noise tolerance in silence.mp3:

        ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -

silenceremove

Remove silence from the beginning, middle or end of the audio.

The filter accepts the following options:

start_periods: This value is used to indicate if audio should be trimmed at beginning of the audio. A value of zero indicates no silence should be trimmed from the beginning. When specifying a non-zero value, it trims audio up until it finds non-silence. Normally, when trimming silence from beginning of audio the start_periods will be 1 but it can be increased to higher values to trim all audio up to specific count of non-silence periods. Default value is 0.
start_duration: Specify the amount of time that non-silence must be detected before it stops trimming audio. By increasing the duration, bursts of noises can be treated as silence and trimmed off. Default value is 0.
start_threshold: This indicates what sample value should be treated as silence. For digital audio, a value of 0 may be fine but for audio recorded from analog, you may wish to increase the value to account for background noise. Can be specified in dB (in case ``dB'' is appended to the specified value) or amplitude ratio. Default value is 0.
stop_periods: Set the count for trimming silence from the end of audio. To remove silence from the middle of a file, specify a stop_periods that is negative. This value is then treated as a positive value and is used to indicate the effect should restart processing as specified by start_periods, making it suitable for removing periods of silence in the middle of the audio. Default value is 0.
stop_duration: Specify a duration of silence that must exist before audio is not copied any more. By specifying a higher duration, silence that is wanted can be left in the audio. Default value is 0.
stop_threshold: This is the same as start_threshold but for trimming silence from the end of audio. Can be specified in dB (in case ``dB'' is appended to the specified value) or amplitude ratio. Default value is 0.
leave_silence: This indicates that stop_duration length of audio should be left intact at the beginning of each period of silence. For example, if you want to remove long pauses between words but do not want to remove the pauses completely. Default value is 0.
detection: Set how is silence detected. Can be "rms" or "peak". Second is faster and works better with digital silence which is exactly 0. Default value is "rms".
window: Set ratio used to calculate size of window for detecting silence. Default value is 0.02. Allowed range is from 0 to 10.

Examples

*

The following example shows how this filter can be used to start a recording that does not contain the delay at the start which usually occurs between pressing the record button and the start of the performance:

        silenceremove=1:5:0.02

*

Trim all silence encountered from beginning to end where there is more than 1 second of silence in audio:

        silenceremove=0:0:0:-1:1:-90dB

sofalizer

SOFAlizer uses head-related transfer functions (HRTFs) to create virtual loudspeakers around the user for binaural listening via headphones (audio formats up to 9 channels supported). The HRTFs are stored in SOFA files (see <http://www.sofacoustics.org/> for a database). SOFAlizer is developed at the Acoustics Research Institute (ARI) of the Austrian Academy of Sciences.

To enable compilation of this filter you need to configure FFmpeg with "--enable-netcdf".

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 degreees. 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.

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, rear left and rear right and also with custom gain:

        "sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|RL 135|RR 225:gain=28"

stereotools

This filter has some handy utilities to manage stereo signals, for converting M/S stereo recordings to L/R signal while having control over the parameters or spreading the stereo image of master track.

The filter accepts the following options:

level_in

Set input level before filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.

level_out

Set output level after filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.

balance_in

Set input balance between both channels. Default is 0. Allowed range is from -1 to 1.

balance_out

Set output balance between both channels. Default is 0. Allowed range is from -1 to 1.

softclip

Enable softclipping. Results in analog distortion instead of harsh digital 0dB clipping. Disabled by default.

mutel

Mute the left channel. Disabled by default.

muter

Mute the right channel. Disabled by default.

phasel

Change the phase of the left channel. Disabled by default.

phaser

Change the phase of the right channel. Disabled by default.

mode

Set stereo mode. Available values are:

lr>lr: Left/Right to Left/Right, this is default.
lr>ms: Left/Right to Mid/Side.
ms>lr: Mid/Side to Left/Right.
lr>ll: Left/Right to Left/Left.
lr>rr: Left/Right to Right/Right.
lr>l+r: Left/Right to Left + Right.
lr>rl: Left/Right to Right/Left.

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.

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.

treble

Boost or cut treble (upper) frequencies of the audio using a two-pole shelving filter with a response similar to that of a standard hi-fi's tone-controls. This is also known as shelving equalisation (EQ).

The filter accepts the following options:

gain, g

Give the gain at whichever is the lower of ~22 kHz and the Nyquist frequency. Its useful range is about -20 (for a large cut) to +20 (for a large boost). Beware of clipping when using a positive gain.

frequency, f

Set the filter's central frequency and so can be used to extend or reduce the frequency range to be boosted or cut. The default value is 3000 Hz.

width_type

Set method to specify band-width of filter.

h: Hz
q: Q-Factor
o: octave
s: slope

width, w

Determine how steep is the filter's shelf transition.

tremolo

Sinusoidal amplitude modulation.

The filter accepts the following options:

f: Modulation frequency in Hertz. Modulation frequencies in the subharmonic range (20 Hz or lower) will result in a tremolo effect. This filter may also be used as a ring modulator by specifying a modulation frequency higher than 20 Hz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.
d: Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.

vibrato

Sinusoidal phase modulation.

The filter accepts the following options:

f: Modulation frequency in Hertz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.
d: Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.

volume

Adjust the input audio volume.

It accepts the following parameters:

volume

Set audio volume expression.

Output values are clipped to the maximum value.

The output audio volume is given by the relation:

        <output_volume> = <volume> * <input_volume>

The default value for volume is ``1.0''.

precision

This parameter represents the mathematical precision.

It determines which input sample formats will be allowed, which affects the precision of the volume scaling.

fixed: 8-bit fixed-point; this limits input sample format to U8, S16, and S32.
float: 32-bit floating-point; this limits input sample format to FLT. (default)
double: 64-bit floating-point; this limits input sample format to DBL.

replaygain

Choose the behaviour on encountering ReplayGain side data in input frames.

drop: Remove ReplayGain side data, ignoring its contents (the default).
ignore: Ignore ReplayGain side data, but leave it in the frame.
track: Prefer the track gain, if present.
album: Prefer the album gain, if present.

replaygain_preamp

Pre-amplification gain in dB to apply to the selected replaygain gain.

Default value for replaygain_preamp is 0.0.

eval

Set when the volume expression is evaluated.

It accepts the following values:

once: only evaluate expression once during the filter initialization, or when the volume command is sent
frame: evaluate expression for each incoming frame

: Default value is once.

The volume expression can contain the following parameters.

n: frame number (starting at zero)
nb_channels: number of channels
nb_consumed_samples: number of samples consumed by the filter
nb_samples: number of samples in the current frame
pos: original frame position in the file
pts: frame PTS
sample_rate: sample rate
startpts: PTS at start of stream
startt: time at start of stream
t: frame time
tb: timestamp timebase
volume: last set volume value

Note that when eval is set to once only the sample_rate and tb variables are available, all other variables will evaluate to NAN.

Commands

This filter supports the following commands:

volume: Modify the volume expression. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
replaygain_noclip: Prevent clipping by limiting the gain applied.
Default value for replaygain_noclip is 1.

Examples

*

Halve the input audio volume:

        volume=volume=0.5
        volume=volume=1/2
        volume=volume=-6.0206dB

In all the above example the named key for volume can be omitted, for example like in:

        volume=0.5

*

Increase input audio power by 6 decibels using fixed-point precision:

        volume=volume=6dB:precision=fixed

*

Fade volume after time 10 with an annihilation period of 5 seconds:

        volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame

volumedetect

Detect the volume of the input video.

The filter has no parameters. The input is not modified. Statistics about the volume will be printed in the log when the input stream end is reached.

In particular it will show the mean volume (root mean square), maximum volume (on a per-sample basis), and the beginning of a histogram of the registered volume values (from the maximum value to a cumulated 1/1000 of the samples).

All volumes are in decibels relative to the maximum PCM value.

Examples

Here is an excerpt of the output:

        [Parsed_volumedetect_0  0xa23120] mean_volume: -27 dB
        [Parsed_volumedetect_0  0xa23120] max_volume: -4 dB
        [Parsed_volumedetect_0  0xa23120] histogram_4db: 6
        [Parsed_volumedetect_0  0xa23120] histogram_5db: 62
        [Parsed_volumedetect_0  0xa23120] histogram_6db: 286
        [Parsed_volumedetect_0  0xa23120] histogram_7db: 1042
        [Parsed_volumedetect_0  0xa23120] histogram_8db: 2551
        [Parsed_volumedetect_0  0xa23120] histogram_9db: 4609
        [Parsed_volumedetect_0  0xa23120] histogram_10db: 8409

It means that:

*: The mean square energy is approximately -27 dB, or 10^-2.7.
*: The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
*: There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

In other words, raising the volume by +4 dB does not cause any clipping, raising it by +5 dB causes clipping for 6 samples, etc.

AUDIO SOURCES

Below is a description of the currently available audio sources.

abuffer

Buffer audio frames, and make them available to the filter chain.

This source is mainly intended for a programmatic use, in particular through the interface defined in libavfilter/asrc_abuffer.h.

It accepts the following parameters:

time_base: The timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form.
sample_rate: The sample rate of the incoming audio buffers.
sample_fmt: The sample format of the incoming audio buffers. Either a sample format name or its corresponding integer representation from the enum AVSampleFormat in libavutil/samplefmt.h
channel_layout: The channel layout of the incoming audio buffers. Either a channel layout name from channel_layout_map in libavutil/channel_layout.c or its corresponding integer representation from the AV_CH_LAYOUT_* macros in libavutil/channel_layout.h
channels: The number of channels of the incoming audio buffers. If both channels and channel_layout are specified, then they must be consistent.

Examples

        abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

will instruct the source to accept planar 16bit signed stereo at 44100Hz. Since the sample format with name ``s16p'' corresponds to the number 6 and the ``stereo'' channel layout corresponds to the value 0x3, this is equivalent to:

        abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

aevalsrc

Generate an audio signal specified by an expression.

This source accepts in input one or more expressions (one for each channel), which are evaluated and used to generate a corresponding audio signal.

This source accepts the following options:

exprs: Set the '|'-separated expressions list for each separate channel. In case the channel_layout option is not specified, the selected channel layout depends on the number of provided expressions. Otherwise the last specified expression is applied to the remaining output channels.
channel_layout, c: Set the channel layout. The number of channels in the specified layout must be equal to the number of specified expressions.
duration, d: Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be greater than the specified duration, as the generated audio is always cut at the end of a complete frame.
If not specified, or the expressed duration is negative, the audio is supposed to be generated forever.
nb_samples, n: Set the number of samples per channel per each output frame, default to 1024.
sample_rate, s: Specify the sample rate, default to 44100.

Each expression in exprs can contain the following constants:

n: number of the evaluated sample, starting from 0
t: time of the evaluated sample expressed in seconds, starting from 0
s: sample rate

Examples

*

Generate silence:

        aevalsrc=0

*

Generate a sin signal with frequency of 440 Hz, set sample rate to 8000 Hz:

        aevalsrc="sin(440*2*PI*t):s=8000"

*

Generate a two channels signal, specify the channel layout (Front Center + Back Center) explicitly:

        aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

*

Generate white noise:

        aevalsrc="-2+random(0)"

*

Generate an amplitude modulated signal:

        aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"

*

Generate 2.5 Hz binaural beats on a 360 Hz carrier:

        aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

anullsrc

The null audio source, return unprocessed audio frames. It is mainly useful as a template and to be employed in analysis / debugging tools, or as the source for filters which ignore the input data (for example the sox synth filter).

This source accepts the following options:

channel_layout, cl: Specifies the channel layout, and can be either an integer or a string representing a channel layout. The default value of channel_layout is ``stereo''.
Check the channel_layout_map definition in libavutil/channel_layout.c for the mapping between strings and channel layout values.
sample_rate, r: Specifies the sample rate, and defaults to 44100.
nb_samples, n: Set the number of samples per requested frames.

Examples

*

Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO.

        anullsrc=r=48000:cl=4

*

Do the same operation with a more obvious syntax:

        anullsrc=r=48000:cl=mono

All the parameters need to be explicitly defined.

flite

Synthesize a voice utterance using the libflite library.

To enable compilation of this filter you need to configure FFmpeg with "--enable-libflite".

Note that the flite library is not thread-safe.

The filter accepts the following options:

list_voices: If set to 1, list the names of the available voices and exit immediately. Default value is 0.
nb_samples, n: Set the maximum number of samples per frame. Default value is 512.
textfile: Set the filename containing the text to speak.
text: Set the text to speak.
voice, v: Set the voice to use for the speech synthesis. Default value is "kal". See also the list_voices option.

Examples

*

Read from file speech.txt, and synthesize the text using the standard flite voice:

        flite=textfile=speech.txt

*

Read the specified text selecting the "slt" voice:

        flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

*

Input text to ffmpeg:

        ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

*

Make ffplay speak the specified text, using "flite" and the "lavfi" device:

        ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'

For more information about libflite, check: <http://www.speech.cs.cmu.edu/flite/>

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, and brown. Default color is white.
seed, s: Specify a value used to seed the PRNG.
nb_samples, n: Set the number of samples per each output frame, default is 1024.

Examples

*

Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of 0.5:

        anoisesrc=d=60:c=pink:r=44100:a=0.5

sine

Generate an audio signal made of a sine wave with amplitude 1/8.

The audio signal is bit-exact.

The filter accepts the following options:

frequency, f

Set the carrier frequency. Default is 440 Hz.

beep_factor, b

Enable a periodic beep every second with frequency beep_factor times the carrier frequency. Default is 0, meaning the beep is disabled.

sample_rate, r

Specify the sample rate, default is 44100.

duration, d

Specify the duration of the generated audio stream.

samples_per_frame

Set the number of samples per output frame.

The expression can contain the following constants:

n: The (sequential) number of the output audio frame, starting from 0.
pts: The PTS (Presentation TimeStamp) of the output audio frame, expressed in TB units.
t: The PTS of the output audio frame, expressed in seconds.
TB: The timebase of the output audio frames.

: Default is 1024.

Examples

*

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.

alphaextract

Extract the alpha component from the input as a grayscale video. This is especially useful with the alphamerge filter.

alphamerge

Add or replace the alpha component of the primary input with the grayscale value of a second input. This is intended for use with alphaextract to allow the transmission or storage of frame sequences that have alpha in a format that doesn't support an alpha channel.

For example, to reconstruct full frames from a normal YUV-encoded video and a separate video created with alphaextract, you might use:

        movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out]

Since this filter is designed for reconstruction, it operates on frame sequences without considering timestamps, and terminates when either input reaches end of stream. This will cause problems if your encoding pipeline drops frames. If you're trying to apply an image as an overlay to a video stream, consider the overlay filter instead.

ass

Same as the subtitles filter, except that it doesn't require libavcodec and libavformat to work. On the other hand, it is limited to ASS (Advanced Substation Alpha) subtitles files.

This filter accepts the following option in addition to the common options from the subtitles filter:

shaping

Set the shaping engine

Available values are:

auto: The default libass shaping engine, which is the best available.
simple: Fast, font-agnostic shaper that can do only substitutions
complex: Slower shaper using OpenType for substitutions and positioning

: The default is "auto".

atadenoise

Apply an Adaptive Temporal Averaging Denoiser to the video input.

The filter accepts the following options:

0a: Set threshold A for 1st plane. Default is 0.02. Valid range is 0 to 0.3.
0b: Set threshold B for 1st plane. Default is 0.04. Valid range is 0 to 5.
1a: Set threshold A for 2nd plane. Default is 0.02. Valid range is 0 to 0.3.
1b: Set threshold B for 2nd plane. Default is 0.04. Valid range is 0 to 5.
2a: Set threshold A for 3rd plane. Default is 0.02. Valid range is 0 to 0.3.
2b: Set threshold B for 3rd plane. Default is 0.04. Valid range is 0 to 5.
Threshold A is designed to react on abrupt changes in the input signal and threshold B is designed to react on continuous changes in the input signal.
s: Set number of frames filter will use for averaging. Default is 33. Must be odd number in range [5, 129].
p: Set what planes of frame filter will use for averaging. Default is all.

avgblur

Apply average blur filter.

The filter accepts the following options:

sizeX: Set horizontal kernel size.
planes: Set which planes to filter. By default all planes are filtered.
sizeY: Set vertical kernel size, if zero it will be same as "sizeX". Default is 0.

bbox

Compute the bounding box for the non-black pixels in the input frame luminance plane.

This filter computes the bounding box containing all the pixels with a luminance value greater than the minimum allowed value. The parameters describing the bounding box are printed on the filter log.

The filter accepts the following option:

min_val: Set the minimal luminance value. Default is 16.

bitplanenoise

Show and measure bit plane noise.

The filter accepts the following options:

bitplane: Set which plane to analyze. Default is 1.
filter: Filter out noisy pixels from "bitplane" set above. Default is disabled.

blackdetect

Detect video intervals that are (almost) completely black. Can be useful to detect chapter transitions, commercials, or invalid recordings. Output lines contains the time for the start, end and duration of the detected black interval expressed in seconds.

In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.

The filter accepts the following options:

black_min_duration, d

Set the minimum detected black duration expressed in seconds. It must be a non-negative floating point number.

Default value is 2.0.

picture_black_ratio_th, pic_th

Set the threshold for considering a picture ``black''. Express the minimum value for the ratio:

        <nb_black_pixels> / <nb_pixels>

for which a picture is considered black. Default value is 0.98.

pixel_black_th, pix_th

Set the threshold for considering a pixel ``black''.

The threshold expresses the maximum pixel luminance value for which a pixel is considered ``black''. The provided value is scaled according to the following equation:

        <absolute_threshold> = <luminance_minimum_value> + <pixel_black_th> * <luminance_range_size>

luminance_range_size and luminance_minimum_value depend on the input video format, the range is [0-255] for YUV full-range formats and [16-235] for YUV non full-range formats.

Default value is 0.10.

The following example sets the maximum pixel threshold to the minimum value, and detects only black intervals of 2 or more seconds:

        blackdetect=d=2:pix_th=0.00

blackframe

Detect frames that are (almost) completely black. Can be useful to detect chapter transitions or commercials. Output lines consist of the frame number of the detected frame, the percentage of blackness, the position in the file if known or -1 and the timestamp in seconds.

In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.

It accepts the following parameters:

amount: The percentage of the pixels that have to be below the threshold; it defaults to 98.
threshold, thresh: The threshold below which a pixel value is considered black; it defaults to 32.

blend, tblend

Blend two video frames into each other.

The "blend" filter takes two input streams and outputs one stream, the first input is the ``top'' layer and second input is ``bottom'' layer. By default, the output terminates when the longest input terminates.

The "tblend" (time blend) filter takes two consecutive frames from one single stream, and outputs the result obtained by blending the new frame on top of the old frame.

A description of the accepted options follows.

c0_mode

c1_mode

c2_mode

c3_mode

all_mode

Set blend mode for specific pixel component or all pixel components in case of all_mode. Default value is "normal".

Available values for component modes are:

addition
addition128
and
average
burn
darken
difference
difference128
divide
dodge
freeze
exclusion
glow
hardlight
hardmix
heat
lighten
linearlight
multiply
multiply128
negation
normal
or
overlay
phoenix
pinlight
reflect
screen
softlight
subtract
vividlight
xor

c0_opacity

c1_opacity

c2_opacity

c3_opacity

all_opacity

Set blend opacity for specific pixel component or all pixel components in case of all_opacity. Only used in combination with pixel component blend modes.

c0_expr

c1_expr

c2_expr

c3_expr

all_expr

Set blend expression for specific pixel component or all pixel components in case of all_expr. Note that related mode options will be ignored if those are set.

The expressions can use the following variables:

N: The sequential number of the filtered frame, starting from 0.
X
Y: the coordinates of the current sample
W
H: the width and height of currently filtered plane
SW
SH: Width and height scale depending on the currently filtered plane. It is the ratio between the corresponding luma plane number of pixels and the current plane ones. E.g. for YUV4:2:0 the values are "1,1" for the luma plane, and "0.5,0.5" for chroma planes.
T: Time of the current frame, expressed in seconds.
TOP, A: Value of pixel component at current location for first video frame (top layer).
BOTTOM, B: Value of pixel component at current location for second video frame (bottom layer).

shortest

Force termination when the shortest input terminates. Default is 0. This option is only defined for the "blend" filter.

repeatlast

Continue applying the last bottom frame after the end of the stream. A value of 0 disable the filter after the last frame of the bottom layer is reached. Default is 1. This option is only defined for the "blend" filter.

Examples

*

Apply transition from bottom layer to top layer in first 10 seconds:

        blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'

*

Apply 1x1 checkerboard effect:

        blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'

*

Apply uncover left effect:

        blend=all_expr='if(gte(N*SW+X,W),A,B)'

*

Apply uncover down effect:

        blend=all_expr='if(gte(Y-N*SH,0),A,B)'

*

Apply uncover up-left effect:

        blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'

*

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=difference128

boxblur

Apply a boxblur algorithm to the input video.

It accepts the following parameters:

luma_radius, lr
luma_power, lp
chroma_radius, cr
chroma_power, cp
alpha_radius, ar
alpha_power, ap

A description of the accepted options follows.

luma_radius, lr

chroma_radius, cr

alpha_radius, ar

Set an expression for the box radius in pixels used for blurring the corresponding input plane.

The radius value must be a non-negative number, and must not be greater than the value of the expression "min(w,h)/2" for the luma and alpha planes, and of "min(cw,ch)/2" for the chroma planes.

Default value for luma_radius is ``2''. If not specified, chroma_radius and alpha_radius default to the corresponding value set for luma_radius.

The expressions can contain the following constants:

w
h: The input width and height in pixels.
cw
ch: The input chroma image width and height in pixels.
hsub
vsub: The horizontal and vertical chroma subsample values. For example, for the pixel format ``yuv422p'', hsub is 2 and vsub is 1.

luma_power, lp

chroma_power, cp

alpha_power, ap

Specify how many times the boxblur filter is applied to the corresponding plane.

Default value for luma_power is 2. If not specified, chroma_power and alpha_power default to the corresponding value set for luma_power.

A value of 0 will disable the effect.

Examples

*

Apply a boxblur filter with the luma, chroma, and alpha radii set to 2:

        boxblur=luma_radius=2:luma_power=1
        boxblur=2:1

*

Set the luma radius to 2, and alpha and chroma radius to 0:

        boxblur=2:1:cr=0:ar=0

*

Set the luma and chroma radii to a fraction of the video dimension:

        boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1

bwdif

Deinterlace the input video (``bwdif'' stands for ``Bob Weaver Deinterlacing Filter'').

Motion adaptive deinterlacing based on yadif with the use of w3fdif and cubic interpolation algorithms. It accepts the following parameters:

mode

The interlacing mode to adopt. It accepts one of the following values:

0, send_frame: Output one frame for each frame.
1, send_field: Output one frame for each field.

: The default value is "send_field".

parity

The picture field parity assumed for the input interlaced video. It accepts one of the following values:

0, tff: Assume the top field is first.
1, bff: Assume the bottom field is first.
-1, auto: Enable automatic detection of field parity.

: The default value is "auto". If the interlacing is unknown or the decoder does not export this information, top field first will be assumed.

deint

Specify which frames to deinterlace. Accept one of the following values:

0, all: Deinterlace all frames.
1, interlaced: Only deinterlace frames marked as interlaced.

: The default value is "all".

chromakey

YUV colorspace color/chroma keying.

The filter accepts the following options:

color

The color which will be replaced with transparency.

similarity

Similarity percentage with the key color.

0.01 matches only the exact key color, while 1.0 matches everything.

blend

Blend percentage.

0.0 makes pixels either fully transparent, or not transparent at all.

Higher values result in semi-transparent pixels, with a higher transparency the more similar the pixels color is to the key color.

yuv

Signals that the color passed is already in YUV instead of RGB.

Litteral colors like ``green'' or ``red'' don't make sense with this enabled anymore. This can be used to pass exact YUV values as hexadecimal numbers.

Examples

*

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

ciescope

Display CIE color diagram with pixels overlaid onto it.

The filter accepts the following options:

system

Set color system.

ntsc, 470m
ebu, 470bg
smpte
240m
apple
widergb
cie1931
rec709, hdtv
uhdtv, rec2020

cie

Set CIE system.

xyy
ucs
luv

gamuts

Set what gamuts to draw.

See "system" option for available values.

size, s

Set ciescope size, by default set to 512.

intensity, i

Set intensity used to map input pixel values to CIE diagram.

contrast

Set contrast used to draw tongue colors that are out of active color system gamut.

corrgamma

Correct gamma displayed on scope, by default enabled.

showwhite

Show white point on CIE diagram, by default disabled.

gamma

Set input gamma. Used only with XYZ input color space.

codecview

Visualize information exported by some codecs.

Some codecs can export information through frames using side-data or other means. For example, some MPEG based codecs export motion vectors through the export_mvs flag in the codec flags2 option.

The filter accepts the following option:

mv

Set motion vectors to visualize.

Available flags for mv are:

pf: forward predicted MVs of P-frames
bf: forward predicted MVs of B-frames
bb: backward predicted MVs of B-frames

qp

Display quantization parameters using the chroma planes.

mv_type, mvt

Set motion vectors type to visualize. Includes MVs from all frames unless specified by frame_type option.

Available flags for mv_type are:

fp: forward predicted MVs
bp: backward predicted MVs

frame_type, ft

Set frame type to visualize motion vectors of.

Available flags for frame_type are:

if: intra-coded frames (I-frames)
pf: predicted frames (P-frames)
bf: bi-directionally predicted frames (B-frames)

Examples

*

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.

Examples

*

Add red color cast to shadows:

        colorbalance=rs=.3

colorkey

RGB colorspace color keying.

The filter accepts the following options:

color

The color which will be replaced with transparency.

similarity

Similarity percentage with the key color.

0.01 matches only the exact key color, while 1.0 matches everything.

blend

Blend percentage.

0.0 makes pixels either fully transparent, or not transparent at all.

Higher values result in semi-transparent pixels, with a higher transparency the more similar the pixels color is to the key color.

Examples

*

Make every green pixel in the input image transparent:

        ffmpeg -i input.png -vf colorkey=green out.png

*

Overlay a greenscreen-video on top of a static background image.

        ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv

colorlevels

Adjust video input frames using levels.

The filter accepts the following options:

rimin
gimin
bimin
aimin: Adjust red, green, blue and alpha input black point. Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.
rimax
gimax
bimax
aimax: Adjust red, green, blue and alpha input white point. Allowed ranges for options are "[-1.0, 1.0]". Defaults are 1.
Input levels are used to lighten highlights (bright tones), darken shadows (dark tones), change the balance of bright and dark tones.
romin
gomin
bomin
aomin: Adjust red, green, blue and alpha output black point. Allowed ranges for options are "[0, 1.0]". Defaults are 0.
romax
gomax
bomax
aomax: Adjust red, green, blue and alpha output white point. Allowed ranges for options are "[0, 1.0]". Defaults are 1.
Output levels allows manual selection of a constrained output level range.

Examples

*

Make video output darker:

        colorlevels=rimin=0.058:gimin=0.058:bimin=0.058

*

Increase contrast:

        colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96

*

Make video output lighter:

        colorlevels=rimax=0.902:gimax=0.902:bimax=0.902

*

Increase brightness:

        colorlevels=romin=0.5:gomin=0.5:bomin=0.5

colorchannelmixer

Adjust video input frames by re-mixing color channels.

This filter modifies a color channel by adding the values associated to the other channels of the same pixels. For example if the value to modify is red, the output value will be:

        <red>=<red>*<rr> + <blue>*<rb> + <green>*<rg> + <alpha>*<ra>

The filter accepts the following options:

rr
rg
rb
ra: Adjust contribution of input red, green, blue and alpha channels for output red channel. Default is 1 for rr, and 0 for rg, rb and ra.
gr
gg
gb
ga: Adjust contribution of input red, green, blue and alpha channels for output green channel. Default is 1 for gg, and 0 for gr, gb and ga.
br
bg
bb
ba: Adjust contribution of input red, green, blue and alpha channels for output blue channel. Default is 1 for bb, and 0 for br, bg and ba.
ar
ag
ab
aa: Adjust contribution of input red, green, blue and alpha channels for output alpha channel. Default is 1 for aa, and 0 for ar, ag and ab.
Allowed ranges for options are "[-2.0, 2.0]".

Examples

*

Convert source to grayscale:

        colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3

*

Simulate sepia tones:

        colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131

colormatrix

Convert color matrix.

The filter accepts the following options:

src

dst

Specify the source and destination color matrix. Both values must be specified.

The accepted values are:

bt709: BT.709
bt601: BT.601
smpte240m: SMPTE-240M
fcc: FCC
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.

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
bt2020ncl: BT.2020 with non-constant luminance

trc

Specify output transfer characteristics.

The accepted values are:

bt709: BT.709
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
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
bt2020: BT.2020

range

Specify output color range.

The accepted values are:

mpeg: MPEG (restricted) range
jpeg: JPEG (full) range

format

Specify output color format.

The accepted values are:

yuv420p: YUV 4:2:0 planar 8-bits
yuv420p10: YUV 4:2:0 planar 10-bits
yuv420p12: YUV 4:2:0 planar 12-bits
yuv422p: YUV 4:2:2 planar 8-bits
yuv422p10: YUV 4:2:2 planar 10-bits
yuv422p12: YUV 4:2:2 planar 12-bits
yuv444p: YUV 4:4:4 planar 8-bits
yuv444p10: YUV 4:4:4 planar 10-bits
yuv444p12: YUV 4:4:4 planar 12-bits

fast

Do a fast conversion, which skips gamma/primary correction. This will take significantly less CPU, but will be mathematically incorrect. To get output compatible with that produced by the colormatrix filter, use fast=1.

dither

Specify dithering mode.

The accepted values are:

none: No dithering
fsb: Floyd-Steinberg dithering

wpadapt

Whitepoint adaptation mode.

The accepted values are:

bradford: Bradford whitepoint adaptation
vonkries: von Kries whitepoint adaptation
identity: identity whitepoint adaptation (i.e. no whitepoint adaptation)

iall

Override all input properties at once. Same accepted values as all.

ispace

Override input colorspace. Same accepted values as space.

iprimaries

Override input color primaries. Same accepted values as primaries.

itrc

Override input transfer characteristics. Same accepted values as trc.

irange

Override input color range. Same accepted values as range.

The filter converts the transfer characteristics, color space and color primaries to the specified user values. The output value, if not specified, is set to a default value based on the ``all'' property. If that property is also not specified, the filter will log an error. The output color range and format default to the same value as the input color range and format. The input transfer characteristics, color space, color primaries and color range should be set on the input data. If any of these are missing, the filter will log an error and no conversion will take place.

For example to convert the input to SMPTE-240M, use the command:

        colorspace=smpte240m

convolution

Apply convolution 3x3 or 5x5 filter.

The filter accepts the following options:

0m
1m
2m
3m: Set matrix for each plane. Matrix is sequence of 9 or 25 signed integers.
0rdiv
1rdiv
2rdiv
3rdiv: Set multiplier for calculated value for each plane.
0bias
1bias
2bias
3bias: Set bias for each plane. This value is added to the result of the multiplication. Useful for making the overall image brighter or darker. Default is 0.0.

Examples

*

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 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"

copy

Copy the input 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

crop

Crop the input video to given dimensions.

It accepts the following parameters:

w, out_w: The width of the output video. It defaults to "iw". This expression is evaluated only once during the filter configuration, or when the w or out_w command is sent.
h, out_h: The height of the output video. It defaults to "ih". This expression is evaluated only once during the filter configuration, or when the h or out_h command is sent.
x: The horizontal position, in the input video, of the left edge of the output video. It defaults to "(in_w-out_w)/2". This expression is evaluated per-frame.
y: The vertical position, in the input video, of the top edge of the output video. It defaults to "(in_h-out_h)/2". This expression is evaluated per-frame.
keep_aspect: If set to 1 will force the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio. It defaults to 0.
exact: Enable exact cropping. If enabled, subsampled videos will be cropped at exact width/height/x/y as specified and will not be rounded to nearest smaller value. It defaults to 0.

The out_w, out_h, x, y parameters are expressions containing the following constants:

x
y: The computed values for x and y. They are evaluated for each new frame.
in_w
in_h: The input width and height.
iw
ih: These are the same as in_w and in_h.
out_w
out_h: The output (cropped) width and height.
ow
oh: These are the same as out_w and out_h.
a: same as iw / ih
sar: input sample aspect ratio
dar: input display aspect ratio, it is the same as (iw / ih) * sar
hsub
vsub: horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.
n: The number of the input frame, starting from 0.
pos: the position in the file of the input frame, NAN if unknown
t: The timestamp expressed in seconds. It's NAN if the input timestamp is unknown.

The expression for out_w may depend on the value of out_h, and the expression for out_h may depend on out_w, but they cannot depend on x and y, as x and y are evaluated after out_w and out_h.

The x and y parameters specify the expressions for the position of the top-left corner of the output (non-cropped) area. They are evaluated for each frame. If the evaluated value is not valid, it is approximated to the nearest valid value.

The expression for x may depend on y, and the expression for y may depend on x.

Examples

*

Crop area with size 100x100 at position (12,34).

        crop=100:100:12:34

Using named options, the example above becomes:

        crop=w=100:h=100:x=12:y=34

*

Crop the central input area with size 100x100:

        crop=100:100

*

Crop the central input area with size 2/3 of the input video:

        crop=2/3*in_w:2/3*in_h

*

Crop the input video central square:

        crop=out_w=in_h
        crop=in_h

*

Delimit the rectangle with the top-left corner placed at position 100:100 and the right-bottom corner corresponding to the right-bottom corner of the input image.

        crop=in_w-100:in_h-100:100:100

*

Crop 10 pixels from the left and right borders, and 20 pixels from the top and bottom borders

        crop=in_w-2*10:in_h-2*20

*

Keep only the bottom right quarter of the input image:

        crop=in_w/2:in_h/2:in_w/2:in_h/2

*

Crop height for getting Greek harmony:

        crop=in_w:1/PHI*in_w

*

Apply trembling effect:

        crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)

*

Apply erratic camera effect depending on timestamp:

        crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)"

*

Set x depending on the value of y:

        crop=in_w/2:in_h/2:y:10+10*sin(n/10)

Commands

This filter supports the following commands:

w, out_w
h, out_h
x
y: Set width/height of the output video and the horizontal/vertical position in the input video. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.

cropdetect

Auto-detect the crop size.

It calculates the necessary cropping parameters and prints the recommended parameters via the logging system. The detected dimensions correspond to the non-black area of the input video.

It accepts the following parameters:

limit: Set higher black value threshold, which can be optionally specified from nothing (0) to everything (255 for 8-bit based formats). An intensity value greater to the set value is considered non-black. It defaults to 24. You can also specify a value between 0.0 and 1.0 which will be scaled depending on the bitdepth of the pixel format.
round: The value which the width/height should be divisible by. It defaults to 16. The offset is automatically adjusted to center the video. Use 2 to get only even dimensions (needed for 4:2:2 video). 16 is best when encoding to most video codecs.
reset_count, reset: Set the counter that determines after how many frames cropdetect will reset the previously detected largest video area and start over to detect the current optimal crop area. Default value is 0.
This can be useful when channel logos distort the video area. 0 indicates 'never reset', and returns the largest area encountered during playback.

curves

Apply color adjustments using curves.

This filter is similar to the Adobe Photoshop and GIMP curves tools. Each component (red, green and blue) has its values defined by N key points tied from each other using a smooth curve. The x-axis represents the pixel values from the input frame, and the y-axis the new pixel values to be set for the output frame.

By default, a component curve is defined by the two points (0;0) and (1;1). This creates a straight line where each original pixel value is ``adjusted'' to its own value, which means no change to the image.

The filter allows you to redefine these two points and add some more. A new curve (using a natural cubic spline interpolation) will be define to pass smoothly through all these new coordinates. The new defined points needs to be strictly increasing over the x-axis, and their x and y values must be in the [0;1] interval. If the computed curves happened to go outside the vector spaces, the values will be clipped accordingly.

The filter accepts the following options:

preset

Select one of the available color presets. This option can be used in addition to the r, g, b parameters; in this case, the later options takes priority on the preset values. Available presets are:

none
color_negative
cross_process
darker
increase_contrast
lighter
linear_contrast
medium_contrast
negative
strong_contrast
vintage

: Default is "none".

master, m

Set the master key points. These points will define a second pass mapping. It is sometimes called a ``luminance'' or ``value'' mapping. It can be used with r, g, b or all since it acts like a post-processing LUT.

red, r

Set the key points for the red component.

green, g

Set the key points for the green component.

blue, b

Set the key points for the blue component.

all

Set the key points for all components (not including master). Can be used in addition to the other key points component options. In this case, the unset component(s) will fallback on this all setting.

psfile

Specify a Photoshop curves file (".acv") to import the settings from.

plot

Save Gnuplot script of the curves in specified file.

To avoid some filtergraph syntax conflicts, each key points list need to be defined using the following syntax: "x0/y0 x1/y1 x2/y2 ...".

Examples

*

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.

dctdnoiz

Denoise frames using 2D DCT (frequency domain filtering).

This filter is not designed for real time.

The filter accepts the following options:

sigma, s

Set the noise sigma constant.

This sigma defines a hard threshold of "3 * sigma"; every DCT coefficient (absolute value) below this threshold with be dropped.

If you need a more advanced filtering, see expr.

Default is 0.

overlap

Set number overlapping pixels for each block. Since the filter can be slow, you may want to reduce this value, at the cost of a less effective filter and the risk of various artefacts.

If the overlapping value doesn't permit processing the whole input width or height, a warning will be displayed and according borders won't be denoised.

Default value is blocksize-1, which is the best possible setting.

expr, e

Set the coefficient factor expression.

For each coefficient of a DCT block, this expression will be evaluated as a multiplier value for the coefficient.

If this is option is set, the sigma option will be ignored.

The absolute value of the coefficient can be accessed through the c variable.

n

Set the blocksize using the number of bits. "1<<n" defines the blocksize, which is the width and height of the processed blocks.

The default value is 3 (8x8) and can be raised to 4 for a blocksize of 16x16. Note that changing this setting has huge consequences on the speed processing. Also, a larger block size does not necessarily means a better de-noising.

Examples

Apply a denoise with a sigma of 4.5:

        dctdnoiz=4.5

The same operation can be achieved using the expression system:

        dctdnoiz=e='gte(c, 4.5*3)'

Violent denoise using a block size of "16x16":

        dctdnoiz=15:n=4

deband

Remove banding artifacts from input video. It works by replacing banded pixels with average value of referenced pixels.

The filter accepts the following options:

1thr
2thr
3thr
4thr: Set banding detection threshold for each plane. Default is 0.02. Valid range is 0.00003 to 0.5. If difference between current pixel and reference pixel is less than threshold, it will be considered as banded.
range, r: Banding detection range in pixels. Default is 16. If positive, random number in range 0 to set value will be used. If negative, exact absolute value will be used. The range defines square of four pixels around current pixel.
direction, d: Set direction in radians from which four pixel will be compared. If positive, random direction from 0 to set direction will be picked. If negative, exact of absolute value will be picked. For example direction 0, -PI or -2*PI radians will pick only pixels on same row and -PI/2 will pick only pixels on same column.
blur: If enabled, current pixel is compared with average value of all four surrounding pixels. The default is enabled. If disabled current pixel is compared with all four surrounding pixels. The pixel is considered banded if only all four differences with surrounding pixels are less than threshold.

decimate

Drop duplicated frames at regular intervals.

The filter accepts the following options:

cycle: Set the number of frames from which one will be dropped. Setting this to N means one frame in every batch of N frames will be dropped. Default is 5.
dupthresh: Set the threshold for duplicate detection. If the difference metric for a frame is less than or equal to this value, then it is declared as duplicate. Default is 1.1
scthresh: Set scene change threshold. Default is 15.
blockx
blocky: Set the size of the x and y-axis blocks used during metric calculations. Larger blocks give better noise suppression, but also give worse detection of small movements. Must be a power of two. Default is 32.
ppsrc: Mark main input as a pre-processed input and activate clean source input stream. This allows the input to be pre-processed with various filters to help the metrics calculation while keeping the frame selection lossless. When set to 1, the first stream is for the pre-processed input, and the second stream is the clean source from where the kept frames are chosen. Default is 0.
chroma: Set whether or not chroma is considered in the metric calculations. Default is 1.

deflate

Apply deflate effect to the video.

This filter replaces the pixel by the local(3x3) average by taking into account only values lower than the pixel.

It accepts the following options:

threshold0
threshold1
threshold2
threshold3: Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.

dejudder

Remove judder produced by partially interlaced telecined content.

Judder can be introduced, for instance, by pullup filter. If the original source was partially telecined content then the output of "pullup,dejudder" will have a variable frame rate. May change the recorded frame rate of the container. Aside from that change, this filter will not affect constant frame rate video.

The option available in this filter is:

cycle

Specify the length of the window over which the judder repeats.

Accepts any integer greater than 1. Useful values are:

4: If the original was telecined from 24 to 30 fps (Film to NTSC).
5: If the original was telecined from 25 to 30 fps (PAL to NTSC).
20: If a mixture of the two.

: The default is 4.

delogo

Suppress a TV station logo by a simple interpolation of the surrounding pixels. Just set a rectangle covering the logo and watch it disappear (and sometimes something even uglier appear - your mileage may vary).

It accepts the following parameters:

x
y: Specify the top left corner coordinates of the logo. They must be specified.
w
h: Specify the width and height of the logo to clear. They must be specified.
band, t: Specify the thickness of the fuzzy edge of the rectangle (added to w and h). The default value is 1. This option is deprecated, setting higher values should no longer be necessary and is not recommended.
show: When set to 1, a green rectangle is drawn on the screen to simplify finding the right x, y, w, and h parameters. The default value is 0.
The rectangle is drawn on the outermost pixels which will be (partly) replaced with interpolated values. The values of the next pixels immediately outside this rectangle in each direction will be used to compute the interpolated pixel values inside the rectangle.

Examples

*

Set a rectangle covering the area with top left corner coordinates 0,0 and size 100x77, and a band of size 10:

        delogo=x=0:y=0:w=100:h=77:band=10

deshake

Attempt to fix small changes in horizontal and/or vertical shift. This filter helps remove camera shake from hand-holding a camera, bumping a tripod, moving on a vehicle, etc.

The filter accepts the following options:

x

y

w

h

Specify a rectangular area where to limit the search for motion vectors. If desired the search for motion vectors can be limited to a rectangular area of the frame defined by its top left corner, width and height. These parameters have the same meaning as the drawbox filter which can be used to visualise the position of the bounding box.

This is useful when simultaneous movement of subjects within the frame might be confused for camera motion by the motion vector search.

If any or all of x, y, w and h are set to -1 then the full frame is used. This allows later options to be set without specifying the bounding box for the motion vector search.

Default - search the whole frame.

rx

ry

Specify the maximum extent of movement in x and y directions in the range 0-64 pixels. Default 16.

edge

Specify how to generate pixels to fill blanks at the edge of the frame. Available values are:

blank, 0: Fill zeroes at blank locations
original, 1: Original image at blank locations
clamp, 2: Extruded edge value at blank locations
mirror, 3: Mirrored edge at blank locations

: Default value is mirror.

blocksize

Specify the blocksize to use for motion search. Range 4-128 pixels, default 8.

contrast

Specify the contrast threshold for blocks. Only blocks with more than the specified contrast (difference between darkest and lightest pixels) will be considered. Range 1-255, default 125.

search

Specify the search strategy. Available values are:

exhaustive, 0: Set exhaustive search
less, 1: Set less exhaustive search.

: Default value is exhaustive.

filename

If set then a detailed log of the motion search is written to the specified file.

opencl

If set to 1, specify using OpenCL capabilities, only available if FFmpeg was configured with "--enable-opencl". Default value is 0.

detelecine

Apply an exact inverse of the telecine operation. It requires a predefined pattern specified using the pattern option which must be the same as that passed to the telecine filter.

This filter accepts the following options:

first_field

top, t: top field first
bottom, b: bottom field first The default value is "top".

pattern

A string of numbers representing the pulldown pattern you wish to apply. The default value is 23.

start_frame

A number representing position of the first frame with respect to the telecine pattern. This is to be used if the stream is cut. The default value is 0.

dilation

Apply dilation effect to the video.

This filter replaces the pixel by the local(3x3) maximum.

It accepts the following options:

threshold0

threshold1

threshold2

threshold3

Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.

coordinates

Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are used.

Flags to local 3x3 coordinates maps like this:

    1 2 3
    4   5
    6 7 8

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.

: 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

drawbox

Draw a colored box on the input image.

It accepts the following parameters:

x
y: The expressions which specify the top left corner coordinates of the box. It defaults to 0.
width, w
height, h: The expressions which specify the width and height of the box; if 0 they are interpreted as the input width and height. It defaults to 0.
color, c: Specify the color of the box to write. For the general syntax of this option, check the ``Color'' section in the ffmpeg-utils manual. If the special value "invert" is used, the box edge color is the same as the video with inverted luma.
thickness, t: The expression which sets the thickness of the box edge. Default value is 3.
See below for the list of accepted constants.

The parameters for x, y, w and h and t are expressions containing the following constants:

dar: The input display aspect ratio, it is the same as (w / h) * sar.
hsub
vsub: horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.
in_h, ih
in_w, iw: The input width and height.
sar: The input sample aspect ratio.
x
y: The x and y offset coordinates where the box is drawn.
w
h: The width and height of the drawn box.
t: The thickness of the drawn box.
These constants allow the x, y, w, h and t expressions to refer to each other, so you may for example specify "y=x/dar" or "h=w/dar".

Examples

*

Draw a black box around the edge of the input image:

        drawbox

*

Draw a box with color red and an opacity of 50%:

        drawbox=10:20:200:60:red@0.5

The previous example can be specified as:

        drawbox=x=10:y=20:w=200:h=60:color=red@0.5

*

Fill the box with pink color:

        drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=max

*

Draw a 2-pixel red 2.40:1 mask:

        drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red

drawgrid

Draw a grid on the input image.

It accepts the following parameters:

x
y: The expressions which specify the coordinates of some point of grid intersection (meant to configure offset). Both default to 0.
width, w
height, h: The expressions which specify the width and height of the grid cell, if 0 they are interpreted as the input width and height, respectively, minus "thickness", so image gets framed. Default to 0.
color, c: Specify the color of the grid. For the general syntax of this option, check the ``Color'' section in the ffmpeg-utils manual. If the special value "invert" is used, the grid color is the same as the video with inverted luma.
thickness, t: The expression which sets the thickness of the grid line. Default value is 1.
See below for the list of accepted constants.

The parameters for x, y, w and h and t are expressions containing the following constants:

dar: The input display aspect ratio, it is the same as (w / h) * sar.
hsub
vsub: horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.
in_h, ih
in_w, iw: The input grid cell width and height.
sar: The input sample aspect ratio.
x
y: The x and y coordinates of some point of grid intersection (meant to configure offset).
w
h: The width and height of the drawn cell.
t: The thickness of the drawn cell.
These constants allow the x, y, w, h and t expressions to refer to each other, so you may for example specify "y=x/dar" or "h=w/dar".

Examples

*

Draw a grid with cell 100x100 pixels, thickness 2 pixels, with color red and an opacity of 50%:

        drawgrid=width=100:height=100:thickness=2:color=red@0.5

*

Draw a white 3x3 grid with an opacity of 50%:

        drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5

drawtext

Draw a text string or text from a specified file on top of a video, using the libfreetype library.

To enable compilation of this filter, you need to configure FFmpeg with "--enable-libfreetype". To enable default font fallback and the font option you need to configure FFmpeg with "--enable-libfontconfig". To enable the text_shaping option, you need to configure FFmpeg with "--enable-libfribidi".

Syntax

It accepts the following parameters:

box

Used to draw a box around text using the background color. The value must be either 1 (enable) or 0 (disable). The default value of box is 0.

boxborderw

Set the width of the border to be drawn around the box using boxcolor. The default value of boxborderw is 0.

boxcolor

The color to be used for drawing box around text. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of boxcolor is ``white''.

borderw

Set the width of the border to be drawn around the text using bordercolor. The default value of borderw is 0.

bordercolor

Set the color to be used for drawing border around text. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of bordercolor is ``black''.

expansion

Select how the text is expanded. Can be either "none", "strftime" (deprecated) or "normal" (default). See the drawtext_expansion, Text expansion section below for details.

fix_bounds

If true, check and fix text coords to avoid clipping.

fontcolor

The color to be used for drawing fonts. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of fontcolor is ``black''.

fontcolor_expr

String which is expanded the same way as text to obtain dynamic fontcolor value. By default this option has empty value and is not processed. When this option is set, it overrides fontcolor option.

font

The font family to be used for drawing text. By default Sans.

fontfile

The font file to be used for drawing text. The path must be included. This parameter is mandatory if the fontconfig support is disabled.

draw

This option does not exist, please see the timeline system

alpha

Draw the text applying alpha blending. The value can be a number between 0.0 and 1.0. The expression accepts the same variables x, y as well. The default value is 1. Please see fontcolor_expr.

fontsize

The font size to be used for drawing text. The default value of fontsize is 16.

text_shaping

If set to 1, attempt to shape the text (for example, reverse the order of right-to-left text and join Arabic characters) before drawing it. Otherwise, just draw the text exactly as given. By default 1 (if supported).

ft_load_flags

The flags to be used for loading the fonts.

The flags map the corresponding flags supported by libfreetype, and are a combination of the following values:

default
no_scale
no_hinting
render
no_bitmap
vertical_layout
force_autohint
crop_bitmap
pedantic
ignore_global_advance_width
no_recurse
ignore_transform
monochrome
linear_design
no_autohint

Default value is ``default''.

For more information consult the documentation for the FT_LOAD_* libfreetype flags.

shadowcolor

The color to be used for drawing a shadow behind the drawn text. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of shadowcolor is ``black''.

shadowx

shadowy

The x and y offsets for the text shadow position with respect to the position of the text. They can be either positive or negative values. The default value for both is ``0''.

start_number

The starting frame number for the n/frame_num variable. The default value is ``0''.

tabsize

The size in number of spaces to use for rendering the tab. Default value is 4.

timecode

Set the initial timecode representation in ``hh:mm:ss[:;.]ff'' format. It can be used with or without text parameter. timecode_rate option must be specified.

timecode_rate, rate, r

Set the timecode frame rate (timecode only).

text

The text string to be drawn. The text must be a sequence of UTF-8 encoded characters. This parameter is mandatory if no file is specified with the parameter textfile.

textfile

A text file containing text to be drawn. The text must be a sequence of UTF-8 encoded characters.

This parameter is mandatory if no text string is specified with the parameter text.

If both text and textfile are specified, an error is thrown.

reload

If set to 1, the textfile will be reloaded before each frame. Be sure to update it atomically, or it may be read partially, or even fail.

x

y

The expressions which specify the offsets where text will be drawn within the video frame. They are relative to the top/left border of the output image.

The default value of x and y is ``0''.

See below for the list of accepted constants and functions.

The parameters for x and y are expressions containing the following constants and functions:

dar: input display aspect ratio, it is the same as (w / h) * sar
hsub
vsub: horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.
line_h, lh: the height of each text line
main_h, h, H: the input height
main_w, w, W: the input width
max_glyph_a, ascent: the maximum distance from the baseline to the highest/upper grid coordinate used to place a glyph outline point, for all the rendered glyphs. It is a positive value, due to the grid's orientation with the Y axis upwards.
max_glyph_d, descent: the maximum distance from the baseline to the lowest grid coordinate used to place a glyph outline point, for all the rendered glyphs. This is a negative value, due to the grid's orientation, with the Y axis upwards.
max_glyph_h: maximum glyph height, that is the maximum height for all the glyphs contained in the rendered text, it is equivalent to ascent - descent.
max_glyph_w: maximum glyph width, that is the maximum width for all the glyphs contained in the rendered text
n: the number of input frame, starting from 0
rand(min, max): return a random number included between min and max
sar: The input sample aspect ratio.
t: timestamp expressed in seconds, NAN if the input timestamp is unknown
text_h, th: the height of the rendered text
text_w, tw: the width of the rendered text
x
y: the x and y offset coordinates where the text is drawn.
These parameters allow the x and y expressions to refer each other, so you can for example specify "y=x/dar".

Text expansion

If expansion is set to "strftime", the filter recognizes strftime() sequences in the provided text and expands them accordingly. Check the documentation of strftime(). This feature is deprecated.

If expansion is set to "none", the text is printed verbatim.

If expansion is set to "normal" (which is the default), the following expansion mechanism is used.

The backslash character \, followed by any character, always expands to the second character.

Sequences of the form "%{...}" are expanded. The text between the braces is a function name, possibly followed by arguments separated by ':'. If the arguments contain special characters or delimiters (':' or '}'), they should be escaped.

Note that they probably must also be escaped as the value for the text option in the filter argument string and as the filter argument in the filtergraph description, and possibly also for the shell, that makes up to four levels of escaping; using a text file avoids these problems.

The following functions are available:

expr, e

The expression evaluation result.

It must take one argument specifying the expression to be evaluated, which accepts the same constants and functions as the x and y values. Note that not all constants should be used, for example the text size is not known when evaluating the expression, so the constants text_w and text_h will have an undefined value.

expr_int_format, eif

Evaluate the expression's value and output as formatted integer.

The first argument is the expression to be evaluated, just as for the expr function. The second argument specifies the output format. Allowed values are x, X, d and u. They are treated exactly as in the "printf" function. The third parameter is optional and sets the number of positions taken by the output. It can be used to add padding with zeros from the left.

gmtime

The time at which the filter is running, expressed in UTC. It can accept an argument: a strftime() format string.

localtime

The time at which the filter is running, expressed in the local time zone. It can accept an argument: a strftime() format string.

metadata

Frame metadata. Takes one or two arguments.

The first argument is mandatory and specifies the metadata key.

The second argument is optional and specifies a default value, used when the metadata key is not found or empty.

n, frame_num

The frame number, starting from 0.

pict_type

A 1 character description of the current picture type.

pts

The timestamp of the current frame. It can take up to three arguments.

The first argument is the format of the timestamp; it defaults to "flt" for seconds as a decimal number with microsecond accuracy; "hms" stands for a formatted [-]HH:MM:SS.mmm timestamp with millisecond accuracy. "gmtime" stands for the timestamp of the frame formatted as UTC time; "localtime" stands for the timestamp of the frame formatted as local time zone time.

The second argument is an offset added to the timestamp.

If the format is set to "localtime" or "gmtime", a third argument may be supplied: a strftime() format string. By default, YYYY-MM-DD HH:MM:SS format will be used.

Examples

*

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'

*

Print the date of a real-time encoding (see strftime(3)):

        drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'

*

Show text fading in and out (appearing/disappearing):

        #!/bin/sh
        DS=1.0 # display start
        DE=10.0 # display end
        FID=1.5 # fade in duration
        FOD=5 # fade out duration
        ffplay -f lavfi "color,drawtext=text=TEST:fontsize=50:fontfile=FreeSerif.ttf:fontcolor_expr=ff0000%{eif\\\\: clip(255*(1*between(t\\, $DS + $FID\\, $DE - $FOD) + ((t - $DS)/$FID)*between(t\\, $DS\\, $DS + $FID) + (-(t - $DE)/$FOD)*between(t\\, $DE - $FOD\\, $DE) )\\, 0\\, 255) \\\\: x\\\\: 2 }"

For more information about libfreetype, check: <http://www.freetype.org/>.

For more information about fontconfig, check: <http://freedesktop.org/software/fontconfig/fontconfig-user.html>.

For more information about libfribidi, check: <http://fribidi.org/>.

edgedetect

Detect and draw edges. The filter uses the Canny Edge Detection algorithm.

The filter accepts the following options:

low

high

Set low and high threshold values used by the Canny thresholding algorithm.

The high threshold selects the ``strong'' edge pixels, which are then connected through 8-connectivity with the ``weak'' edge pixels selected by the low threshold.

low and high threshold values must be chosen in the range [0,1], and low should be lesser or equal to high.

Default value for low is "20/255", and default value for high is "50/255".

mode

Define the drawing mode.

wires: Draw white/gray wires on black background.
colormix: Mix the colors to create a paint/cartoon effect.

: Default value is wires.

Examples

*

Standard edge detection with custom values for the hysteresis thresholding:

        edgedetect=low=0.1:high=0.4

*

Painting effect without thresholding:

        edgedetect=mode=colormix:high=0

eq

Set brightness, contrast, saturation and approximate gamma adjustment.

The filter accepts the following options:

contrast

Set the contrast expression. The value must be a float value in range "-2.0" to 2.0. The default value is ``1''.

brightness

Set the brightness expression. The value must be a float value in range "-1.0" to 1.0. The default value is ``0''.

saturation

Set the saturation expression. The value must be a float in range 0.0 to 3.0. The default value is ``1''.

gamma

Set the gamma expression. The value must be a float in range 0.1 to 10.0. The default value is ``1''.

gamma_r

Set the gamma expression for red. The value must be a float in range 0.1 to 10.0. The default value is ``1''.

gamma_g

Set the gamma expression for green. The value must be a float in range 0.1 to 10.0. The default value is ``1''.

gamma_b

Set the gamma expression for blue. The value must be a float in range 0.1 to 10.0. The default value is ``1''.

gamma_weight

Set the gamma weight expression. It can be used to reduce the effect of a high gamma value on bright image areas, e.g. keep them from getting overamplified and just plain white. The value must be a float in range 0.0 to 1.0. A value of 0.0 turns the gamma correction all the way down while 1.0 leaves it at its full strength. Default is ``1''.

eval

Set when the expressions for brightness, contrast, saturation and gamma expressions are evaluated.

It accepts the following values:

init: only evaluate expressions once during the filter initialization or when a command is processed
frame: evaluate expressions for each incoming frame

: Default value is init.

The expressions accept the following parameters:

n: frame count of the input frame starting from 0
pos: byte position of the corresponding packet in the input file, NAN if unspecified
r: frame rate of the input video, NAN if the input frame rate is unknown
t: timestamp expressed in seconds, NAN if the input timestamp is unknown

Commands

The filter supports the following commands:

contrast

Set the contrast expression.

brightness

Set the brightness expression.

saturation

Set the saturation expression.

gamma

Set the gamma expression.

gamma_r

Set the gamma_r expression.

gamma_g

Set gamma_g expression.

gamma_b

Set gamma_b expression.

gamma_weight

Set gamma_weight expression.

The command accepts the same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current value.

erosion

Apply erosion effect to the video.

This filter replaces the pixel by the local(3x3) minimum.

It accepts the following options:

threshold0

threshold1

threshold2

threshold3

Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.

coordinates

Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are used.

Flags to local 3x3 coordinates maps like this:

    1 2 3
    4   5
    6 7 8

extractplanes

Extract color channel components from input video stream into separate grayscale video streams.

The filter accepts the following option:

planes

Set plane(s) to extract.

Available values for planes are:

y
u
v
a
r
g
b

: Choosing planes not available in the input will result in an error. That means you cannot select "r", "g", "b" planes with "y", "u", "v" planes at same time.

Examples

*

Extract luma, u and v color channel component from input video frame into 3 grayscale outputs:

        ffmpeg -i video.avi -filter_complex 'extractplanes=y+u+v[y][u][v]' -map '[y]' y.avi -map '[u]' u.avi -map '[v]' v.avi

elbg

Apply a posterize effect using the ELBG (Enhanced LBG) algorithm.

For each input image, the filter will compute the optimal mapping from the input to the output given the codebook length, that is the number of distinct output colors.

This filter accepts the following options.

codebook_length, l: Set codebook length. The value must be a positive integer, and represents the number of distinct output colors. Default value is 256.
nb_steps, n: Set the maximum number of iterations to apply for computing the optimal mapping. The higher the value the better the result and the higher the computation time. Default value is 1.
seed, s: Set a random seed, must be an integer included between 0 and UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random seed on a best effort basis.
pal8: Set pal8 output pixel format. This option does not work with codebook length greater than 256.

fade

Apply a fade-in/out effect to the input video.

It accepts the following parameters:

type, t: The effect type can be either ``in'' for a fade-in, or ``out'' for a fade-out effect. Default is "in".
start_frame, s: Specify the number of the frame to start applying the fade effect at. Default is 0.
nb_frames, n: The number of frames that the fade effect lasts. At the end of the fade-in effect, the output video will have the same intensity as the input video. At the end of the fade-out transition, the output video will be filled with the selected color. Default is 25.
alpha: If set to 1, fade only alpha channel, if one exists on the input. Default value is 0.
start_time, st: Specify the timestamp (in seconds) of the frame to start to apply the fade effect. If both start_frame and start_time are specified, the fade will start at whichever comes last. Default is 0.
duration, d: The number of seconds for which the fade effect has to last. At the end of the fade-in effect the output video will have the same intensity as the input video, at the end of the fade-out transition the output video will be filled with the selected color. If both duration and nb_frames are specified, duration is used. Default is 0 (nb_frames is used by default).
color, c: Specify the color of the fade. Default is ``black''.

Examples

*

Fade in the first 30 frames of video:

        fade=in:0:30

The command above is equivalent to:

        fade=t=in:s=0:n=30

*

Fade out the last 45 frames of a 200-frame video:

        fade=out:155:45
        fade=type=out:start_frame=155:nb_frames=45

*

Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video:

        fade=in:0:25, fade=out:975:25

*

Make the first 5 frames yellow, then fade in from frame 5-24:

        fade=in:5:20:color=yellow

*

Fade in alpha over first 25 frames of video:

        fade=in:0:25:alpha=1

*

Make the first 5.5 seconds black, then fade in for 0.5 seconds:

        fade=t=in:st=5.5:d=0.5

fftfilt

Apply arbitrary expressions to samples in frequency domain

dc_Y: Adjust the dc value (gain) of the luma plane of the image. The filter accepts an integer value in range 0 to 1000. The default value is set to 0.
dc_U: Adjust the dc value (gain) of the 1st chroma plane of the image. The filter accepts an integer value in range 0 to 1000. The default value is set to 0.
dc_V: Adjust the dc value (gain) of the 2nd chroma plane of the image. The filter accepts an integer value in range 0 to 1000. The default value is set to 0.
weight_Y: Set the frequency domain weight expression for the luma plane.
weight_U: Set the frequency domain weight expression for the 1st chroma plane.
weight_V: Set the frequency domain weight expression for the 2nd chroma plane.
The filter accepts the following variables:
X
Y: The coordinates of the current sample.
W
H: The width and height of the image.

Examples

*

High-pass:

        fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'

*

Low-pass:

        fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'

*

Sharpen:

        fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'

*

Blur:

        fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))'

field

Extract a single field from an interlaced image using stride arithmetic to avoid wasting CPU time. The output frames are marked as non-interlaced.

The filter accepts the following options:

type: Specify whether to extract the top (if the value is 0 or "top") or the bottom field (if the value is 1 or "bottom").

fieldhint

Create new frames by copying the top and bottom fields from surrounding frames supplied as numbers by the hint file.

hint

Set file containing hints: absolute/relative frame numbers.

There must be one line for each frame in a clip. Each line must contain two numbers separated by the comma, optionally followed by "-" or "+". Numbers supplied on each line of file can not be out of [N-1,N+1] where N is current frame number for "absolute" mode or out of [-1, 1] range for "relative" mode. First number tells from which frame to pick up top field and second number tells from which frame to pick up bottom field.

If optionally followed by "+" output frame will be marked as interlaced, else if followed by "-" output frame will be marked as progressive, else it will be marked same as input frame. If line starts with "#" or ";" that line is skipped.

mode

Can be item "absolute" or "relative". Default is "absolute".

Example of first several lines of "hint" file for "relative" mode:

        0,0 - # first frame
        1,0 - # second frame, use third's frame top field and second's frame bottom field
        1,0 - # third frame, use fourth's frame top field and third's frame bottom field
        1,0 -
        0,0 -
        0,0 -
        1,0 -
        1,0 -
        1,0 -
        0,0 -
        0,0 -
        1,0 -
        1,0 -
        1,0 -
        0,0 -

fieldmatch

Field matching filter for inverse telecine. It is meant to reconstruct the progressive frames from a telecined stream. The filter does not drop duplicated frames, so to achieve a complete inverse telecine "fieldmatch" needs to be followed by a decimation filter such as decimate in the filtergraph.

The separation of the field matching and the decimation is notably motivated by the possibility of inserting a de-interlacing filter fallback between the two. If the source has mixed telecined and real interlaced content, "fieldmatch" will not be able to match fields for the interlaced parts. But these remaining combed frames will be marked as interlaced, and thus can be de-interlaced by a later filter such as yadif before decimation.

In addition to the various configuration options, "fieldmatch" can take an optional second stream, activated through the ppsrc option. If enabled, the frames reconstruction will be based on the fields and frames from this second stream. This allows the first input to be pre-processed in order to help the various algorithms of the filter, while keeping the output lossless (assuming the fields are matched properly). Typically, a field-aware denoiser, or brightness/contrast adjustments can help.

Note that this filter uses the same algorithms as TIVTC/TFM (AviSynth project) and VIVTC/VFM (VapourSynth project). The later is a light clone of TFM from which "fieldmatch" is based on. While the semantic and usage are very close, some behaviour and options names can differ.

The decimate filter currently only works for constant frame rate input. If your input has mixed telecined (30fps) and progressive content with a lower framerate like 24fps use the following filterchain to produce the necessary cfr stream: "dejudder,fps=30000/1001,fieldmatch,decimate".

The filter accepts the following options:

order

Specify the assumed field order of the input stream. Available values are:

auto: Auto detect parity (use FFmpeg's internal parity value).
bff: Assume bottom field first.
tff: Assume top field first.

Note that it is sometimes recommended not to trust the parity announced by the stream.

Default value is auto.

mode

Set the matching mode or strategy to use. pc mode is the safest in the sense that it won't risk creating jerkiness due to duplicate frames when possible, but if there are bad edits or blended fields it will end up outputting combed frames when a good match might actually exist. On the other hand, pcn_ub mode is the most risky in terms of creating jerkiness, but will almost always find a good frame if there is one. The other values are all somewhere in between pc and pcn_ub in terms of risking jerkiness and creating duplicate frames versus finding good matches in sections with bad edits, orphaned fields, blended fields, etc.

More details about p/c/n/u/b are available in p/c/n/u/b meaning section.

Available values are:

pc: 2-way matching (p/c)
pc_n: 2-way matching, and trying 3rd match if still combed (p/c + n)
pc_u: 2-way matching, and trying 3rd match (same order) if still combed (p/c + u)
pc_n_ub: 2-way matching, trying 3rd match if still combed, and trying 4th/5th matches if still combed (p/c + n + u/b)
pcn: 3-way matching (p/c/n)
pcn_ub: 3-way matching, and trying 4th/5th matches if all 3 of the original matches are detected as combed (p/c/n + u/b)

The parenthesis at the end indicate the matches that would be used for that mode assuming order=tff (and field on auto or top).

In terms of speed pc mode is by far the fastest and pcn_ub is the slowest.

Default value is pc_n.

ppsrc

Mark the main input stream as a pre-processed input, and enable the secondary input stream as the clean source to pick the fields from. See the filter introduction for more details. It is similar to the clip2 feature from VFM/TFM.

Default value is 0 (disabled).

field

Set the field to match from. It is recommended to set this to the same value as order unless you experience matching failures with that setting. In certain circumstances changing the field that is used to match from can have a large impact on matching performance. Available values are:

auto: Automatic (same value as order).
bottom: Match from the bottom field.
top: Match from the top field.

: Default value is auto.

mchroma

Set whether or not chroma is included during the match comparisons. In most cases it is recommended to leave this enabled. You should set this to 0 only if your clip has bad chroma problems such as heavy rainbowing or other artifacts. Setting this to 0 could also be used to speed things up at the cost of some accuracy.

Default value is 1.

y0

y1

These define an exclusion band which excludes the lines between y0 and y1 from being included in the field matching decision. An exclusion band can be used to ignore subtitles, a logo, or other things that may interfere with the matching. y0 sets the starting scan line and y1 sets the ending line; all lines in between y0 and y1 (including y0 and y1) will be ignored. Setting y0 and y1 to the same value will disable the feature. y0 and y1 defaults to 0.

scthresh

Set the scene change detection threshold as a percentage of maximum change on the luma plane. Good values are in the "[8.0, 14.0]" range. Scene change detection is only relevant in case combmatch=sc. The range for scthresh is "[0.0, 100.0]".

Default value is 12.0.

combmatch

When combatch is not none, "fieldmatch" will take into account the combed scores of matches when deciding what match to use as the final match. Available values are:

none: No final matching based on combed scores.
sc: Combed scores are only used when a scene change is detected.
full: Use combed scores all the time.

: Default is sc.

combdbg

Force "fieldmatch" to calculate the combed metrics for certain matches and print them. This setting is known as micout in TFM/VFM vocabulary. Available values are:

none: No forced calculation.
pcn: Force p/c/n calculations.
pcnub: Force p/c/n/u/b calculations.

: Default value is none.

cthresh

This is the area combing threshold used for combed frame detection. This essentially controls how ``strong'' or ``visible'' combing must be to be detected. Larger values mean combing must be more visible and smaller values mean combing can be less visible or strong and still be detected. Valid settings are from "-1" (every pixel will be detected as combed) to 255 (no pixel will be detected as combed). This is basically a pixel difference value. A good range is "[8, 12]".

Default value is 9.

chroma

Sets whether or not chroma is considered in the combed frame decision. Only disable this if your source has chroma problems (rainbowing, etc.) that are causing problems for the combed frame detection with chroma enabled. Actually, using chroma=0 is usually more reliable, except for the case where there is chroma only combing in the source.

Default value is 0.

blockx

blocky

Respectively set the x-axis and y-axis size of the window used during combed frame detection. This has to do with the size of the area in which combpel pixels are required to be detected as combed for a frame to be declared combed. See the combpel parameter description for more info. Possible values are any number that is a power of 2 starting at 4 and going up to 512.

Default value is 16.

combpel

The number of combed pixels inside any of the blocky by blockx size blocks on the frame for the frame to be detected as combed. While cthresh controls how ``visible'' the combing must be, this setting controls ``how much'' combing there must be in any localized area (a window defined by the blockx and blocky settings) on the frame. Minimum value is 0 and maximum is "blocky x blockx" (at which point no frames will ever be detected as combed). This setting is known as MI in TFM/VFM vocabulary.

Default value is 80.

p/c/n/u/b meaning

p/c/n

We assume the following telecined stream:

        Top fields:     1 2 2 3 4
        Bottom fields:  1 2 3 4 4

The numbers correspond to the progressive frame the fields relate to. Here, the first two frames are progressive, the 3rd and 4th are combed, and so on.

When "fieldmatch" is configured to run a matching from bottom (field=bottom) this is how this input stream get transformed:

        Input stream:
                        T     1 2 2 3 4
                        B     1 2 3 4 4   <-- matching reference
        
        Matches:              c c n n c
        
        Output stream:
                        T     1 2 3 4 4
                        B     1 2 3 4 4

As a result of the field matching, we can see that some frames get duplicated. To perform a complete inverse telecine, you need to rely on a decimation filter after this operation. See for instance the decimate filter.

The same operation now matching from top fields (field=top) looks like this:

        Input stream:
                        T     1 2 2 3 4   <-- matching reference
                        B     1 2 3 4 4
        
        Matches:              c c p p c
        
        Output stream:
                        T     1 2 2 3 4
                        B     1 2 2 3 4

In these examples, we can see what p, c and n mean; basically, they refer to the frame and field of the opposite parity:

*<p matches the field of the opposite parity in the previous frame>
*<c matches the field of the opposite parity in the current frame>
*<n matches the field of the opposite parity in the next frame>

u/b

The u and b matching are a bit special in the sense that they match from the opposite parity flag. In the following examples, we assume that we are currently matching the 2nd frame (Top:2, bottom:2). According to the match, a 'x' is placed above and below each matched fields.

With bottom matching (field=bottom):

        Match:           c         p           n          b          u
        
                         x       x               x        x          x
          Top          1 2 2     1 2 2       1 2 2      1 2 2      1 2 2
          Bottom       1 2 3     1 2 3       1 2 3      1 2 3      1 2 3
                         x         x           x        x              x
        
        Output frames:
                         2          1          2          2          2
                         2          2          2          1          3

With top matching (field=top):

        Match:           c         p           n          b          u
        
                         x         x           x        x              x
          Top          1 2 2     1 2 2       1 2 2      1 2 2      1 2 2
          Bottom       1 2 3     1 2 3       1 2 3      1 2 3      1 2 3
                         x       x               x        x          x
        
        Output frames:
                         2          2          2          1          2
                         2          1          3          2          2

Examples

Simple IVTC of a top field first telecined stream:

        fieldmatch=order=tff:combmatch=none, decimate

Advanced IVTC, with fallback on yadif for still combed frames:

        fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate

fieldorder

Transform the field order of the input video.

It accepts the following parameters:

order: The output field order. Valid values are tff for top field first or bff for bottom field first.

The default value is tff.

The transformation is done by shifting the picture content up or down by one line, and filling the remaining line with appropriate picture content. This method is consistent with most broadcast field order converters.

If the input video is not flagged as being interlaced, or it is already flagged as being of the required output field order, then this filter does not alter the incoming video.

It is very useful when converting to or from PAL DV material, which is bottom field first.

For example:

        ffmpeg -i in.vob -vf "fieldorder=bff" out.dv

fifo, afifo

Buffer input images and send them when they are requested.

It is mainly useful when auto-inserted by the libavfilter framework.

It does not take parameters.

find_rect

Find a rectangular object

It accepts the following options:

object: Filepath of the object image, needs to be in gray8.
threshold: Detection threshold, default is 0.5.
mipmaps: Number of mipmaps, default is 3.
xmin, ymin, xmax, ymax: Specifies the rectangle in which to search.

Examples

*

Generate a representative palette of a given video using ffmpeg:

        ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

cover_rect

Cover a rectangular object

It accepts the following options:

cover

Filepath of the optional cover image, needs to be in yuv420.

mode

Set covering mode.

It accepts the following values:

cover: cover it by the supplied image
blur: cover it by interpolating the surrounding pixels

: Default value is blur.

Examples

*

Generate a representative palette of a given video using ffmpeg:

        ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

format

Convert the input video to one of the specified pixel formats. Libavfilter will try to pick one that is suitable as input to the next filter.

It accepts the following parameters:

pix_fmts: A '|'-separated list of pixel format names, such as ``pix_fmts=yuv420p|monow|rgb24''.

Examples

*

Convert the input video to the yuv420p format

        format=pix_fmts=yuv420p

Convert the input video to any of the formats in the list

        format=pix_fmts=yuv420p|yuv444p|yuv410p

fps

Convert the video to specified constant frame rate by duplicating or dropping frames as necessary.

It accepts the following parameters:

fps

The desired output frame rate. The default is 25.

round

Rounding method.

Possible values are:

zero: zero round towards 0
inf: round away from 0
down: round towards -infinity
up: round towards +infinity
near: round to nearest

: The default is "near".

start_time

Assume the first PTS should be the given value, in seconds. This allows for padding/trimming at the start of stream. By default, no assumption is made about the first frame's expected PTS, so no padding or trimming is done. For example, this could be set to 0 to pad the beginning with duplicates of the first frame if a video stream starts after the audio stream or to trim any frames with a negative PTS.

Alternatively, the options can be specified as a flat string: fps[:round].

framepack

Pack two different video streams into a stereoscopic video, setting proper metadata on supported codecs. The two views should have the same size and framerate and processing will stop when the shorter video ends. Please note that you may conveniently adjust view properties with the scale and fps filters.

It accepts the following parameters:

format

The desired packing format. Supported values are:

sbs: The views are next to each other (default).
tab: The views are on top of each other.
lines: The views are packed by line.
columns: The views are packed by column.
frameseq: The views are temporally interleaved.

Some examples:

        # Convert left and right views into a frame-sequential video
        ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT
        
        # Convert views into a side-by-side video with the same output resolution as the input
        ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

framerate

Change the frame rate by interpolating new video output frames from the source frames.

This filter is not designed to function correctly with interlaced media. If you wish to change the frame rate of interlaced media then you are required to deinterlace before this filter and re-interlace after this filter.

A description of the accepted options follows.

fps

Specify the output frames per second. This option can also be specified as a value alone. The default is 50.

interp_start

Specify the start of a range where the output frame will be created as a linear interpolation of two frames. The range is [0-255], the default is 15.

interp_end

Specify the end of a range where the output frame will be created as a linear interpolation of two frames. The range is [0-255], the default is 240.

scene

Specify the level at which a scene change is detected as a value between 0 and 100 to indicate a new scene; a low value reflects a low probability for the current frame to introduce a new scene, while a higher value means the current frame is more likely to be one. The default is 7.

flags

Specify flags influencing the filter process.

Available value for flags is:

scene_change_detect, scd: Enable scene change detection using the value of the option scene. This flag is enabled by default.

framestep

Select one frame every N-th frame.

This filter accepts the following option:

step: Select frame after every "step" frames. Allowed values are positive integers higher than 0. Default value is 1.

frei0r

Apply a frei0r effect to the input video.

To enable the compilation of this filter, you need to install the frei0r header and configure FFmpeg with "--enable-frei0r".

It accepts the following parameters:

filter_name: The name of the frei0r effect to load. If the environment variable FREI0R_PATH is defined, the frei0r effect is searched for in each of the directories specified by the colon-separated list in FREIOR_PATH. Otherwise, the standard frei0r paths are searched, in this order: HOME/.frei0r-1/lib/, /usr/local/lib/frei0r-1/, /usr/lib/frei0r-1/.
filter_params: A '|'-separated list of parameters to pass to the frei0r effect.

A frei0r effect parameter can be a boolean (its value is either ``y'' or ``n''), a double, a color (specified as R/G/B, where R, G, and B are floating point numbers between 0.0 and 1.0, inclusive) or by a color description specified in the ``Color'' section in the ffmpeg-utils manual), a position (specified as X/Y, where X and Y are floating point numbers) and/or a string.

The number and types of parameters depend on the loaded effect. If an effect parameter is not specified, the default value is set.

Examples

*

Apply the distort0r effect, setting the first two double parameters:

        frei0r=filter_name=distort0r:filter_params=0.5|0.01

*

Apply the colordistance effect, taking a color as the first parameter:

        frei0r=colordistance:0.2/0.3/0.4
        frei0r=colordistance:violet
        frei0r=colordistance:0x112233

*

Apply the perspective effect, specifying the top left and top right image positions:

        frei0r=perspective:0.2/0.2|0.8/0.2

For more information, see <http://frei0r.dyne.org>

fspp

Apply fast and simple postprocessing. It is a faster version of spp.

It splits (I)DCT into horizontal/vertical passes. Unlike the simple post- processing filter, one of them is performed once per block, not per pixel. This allows for much higher speed.

The filter accepts the following options:

quality: Set quality. This option defines the number of levels for averaging. It accepts an integer in the range 4-5. Default value is 4.
qp: Force a constant quantization parameter. It accepts an integer in range 0-63. If not set, the filter will use the QP from the video stream (if available).
strength: Set filter strength. It accepts an integer in range -15 to 32. Lower values mean more details but also more artifacts, while higher values make the image smoother but also blurrier. Default value is 0 X PSNR optimal.
use_bframe_qp: Enable the use of the QP from the B-Frames if set to 1. Using this option may cause flicker since the B-Frames have often larger QP. Default is 0 (not enabled).

gblur

Apply Gaussian blur filter.

The filter accepts the following options:

sigma: Set horizontal sigma, standard deviation of Gaussian blur. Default is 0.5.
steps: Set number of steps for Gaussian approximation. Defauls is 1.
planes: Set which planes to filter. By default all planes are filtered.
sigmaV: Set vertical sigma, if negative it will be same as "sigma". Default is "-1".

geq

The filter accepts the following options:

lum_expr, lum: Set the luminance expression.
cb_expr, cb: Set the chrominance blue expression.
cr_expr, cr: Set the chrominance red expression.
alpha_expr, a: Set the alpha expression.
red_expr, r: Set the red expression.
green_expr, g: Set the green expression.
blue_expr, b: Set the blue expression.

The colorspace is selected according to the specified options. If one of the lum_expr, cb_expr, or cr_expr options is specified, the filter will automatically select a YCbCr colorspace. If one of the red_expr, green_expr, or blue_expr options is specified, it will select an RGB colorspace.

If one of the chrominance expression is not defined, it falls back on the other one. If no alpha expression is specified it will evaluate to opaque value. If none of chrominance expressions are specified, they will evaluate to the luminance expression.

The expressions can use the following variables and functions:

N: The sequential number of the filtered frame, starting from 0.
X
Y: The coordinates of the current sample.
W
H: The width and height of the image.
SW
SH: Width and height scale depending on the currently filtered plane. It is the ratio between the corresponding luma plane number of pixels and the current plane ones. E.g. for YUV4:2:0 the values are "1,1" for the luma plane, and "0.5,0.5" for chroma planes.
T: Time of the current frame, expressed in seconds.
p(x, y): Return the value of the pixel at location (x,y) of the current plane.
lum(x, y): Return the value of the pixel at location (x,y) of the luminance plane.
cb(x, y): Return the value of the pixel at location (x,y) of the blue-difference chroma plane. Return 0 if there is no such plane.
cr(x, y): Return the value of the pixel at location (x,y) of the red-difference chroma plane. Return 0 if there is no such plane.
r(x, y)
g(x, y)
b(x, y): Return the value of the pixel at location (x,y) of the red/green/blue component. Return 0 if there is no such component.
alpha(x, y): Return the value of the pixel at location (x,y) of the alpha plane. Return 0 if there is no such plane.

For functions, if x and y are outside the area, the value will be automatically clipped to the closer edge.

Examples

*

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

haldclut

Apply a Hald CLUT to a video stream.

First input is the video stream to process, and second one is the Hald CLUT. The Hald CLUT input can be a simple picture or a complete video stream.

The filter accepts the following options:

shortest: Force termination when the shortest input terminates. Default is 0.
repeatlast: Continue applying the last CLUT after the end of the stream. A value of 0 disable the filter after the last frame of the CLUT is reached. Default is 1.

"haldclut" also has the same interpolation options as lut3d (both filters share the same internals).

More information about the Hald CLUT can be found on Eskil Steenberg's website (Hald CLUT author) at <http://www.quelsolaar.com/technology/clut.html>.

Workflow examples

Hald CLUT video stream

Generate an identity Hald CLUT stream altered with various effects:

        ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut

Note: make sure you use a lossless codec.

Then use it with "haldclut" to apply it on some random stream:

        ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv

The Hald CLUT will be applied to the 10 first seconds (duration of clut.nut), then the latest picture of that CLUT stream will be applied to the remaining frames of the "mandelbrot" stream.

Hald CLUT with preview

A Hald CLUT is supposed to be a squared image of "Level*Level*Level" by "Level*Level*Level" pixels. For a given Hald CLUT, FFmpeg will select the biggest possible square starting at the top left of the picture. The remaining padding pixels (bottom or right) will be ignored. This area can be used to add a preview of the Hald CLUT.

Typically, the following generated Hald CLUT will be supported by the "haldclut" filter:

        ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "
           pad=iw+320 [padded_clut];
           smptebars=s=320x256, split [a][b];
           [padded_clut][a] overlay=W-320:h, curves=color_negative [main];
           [main][b] overlay=W-320" -frames:v 1 clut.png

It contains the original and a preview of the effect of the CLUT: SMPTE color bars are displayed on the right-top, and below the same color bars processed by the color changes.

Then, the effect of this Hald CLUT can be visualized with:

        ffplay input.mkv -vf "movie=clut.png, [in] haldclut"

hflip

Flip the input video horizontally.

For example, to horizontally flip the input video with ffmpeg:

        ffmpeg -i in.avi -vf "hflip" out.avi

histeq

This filter applies a global color histogram equalization on a per-frame basis.

It can be used to correct video that has a compressed range of pixel intensities. The filter redistributes the pixel intensities to equalize their distribution across the intensity range. It may be viewed as an ``automatically adjusting contrast filter''. This filter is useful only for correcting degraded or poorly captured source video.

The filter accepts the following options:

strength: Determine the amount of equalization to be applied. As the strength is reduced, the distribution of pixel intensities more-and-more approaches that of the input frame. The value must be a float number in the range [0,1] and defaults to 0.200.
intensity: Set the maximum intensity that can generated and scale the output values appropriately. The strength should be set as desired and then the intensity can be limited if needed to avoid washing-out. The value must be a float number in the range [0,1] and defaults to 0.210.
antibanding: Set the antibanding level. If enabled the filter will randomly vary the luminance of output pixels by a small amount to avoid banding of the histogram. Possible values are "none", "weak" or "strong". It defaults to "none".

histogram

Compute and draw a color distribution histogram for the input video.

The computed histogram is a representation of the color component distribution in an image.

Standard histogram displays the color components distribution in an image. Displays color graph for each color component. Shows distribution of the Y, U, V, A or R, G, B components, depending on input format, in the current frame. Below each graph a color component scale meter is shown.

The filter accepts the following options:

level_height

Set height of level. Default value is 200. Allowed range is [50, 2048].

scale_height

Set height of color scale. Default value is 12. Allowed range is [0, 40].

display_mode

Set display mode. It accepts the following values:

parade: Per color component graphs are placed below each other.
overlay: Presents information identical to that in the "parade", except that the graphs representing color components are superimposed directly over one another.

: Default is "parade".

levels_mode

Set mode. Can be either "linear", or "logarithmic". Default is "linear".

components

Set what color components to display. Default is 7.

fgopacity

Set foreground opacity. Default is 0.7.

bgopacity

Set background opacity. Default is 0.5.

Examples

*

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.

hwupload_cuda

Upload system memory frames to a CUDA device.

It accepts the following optional parameters:

device: The number of the CUDA device to use

hqx

Apply a high-quality magnification filter designed for pixel art. This filter was originally created by Maxim Stepin.

It accepts the following option:

n: Set the scaling dimension: 2 for "hq2x", 3 for "hq3x" and 4 for "hq4x". Default is 3.

hstack

Stack input videos horizontally.

All streams must be of same pixel format and of same height.

Note that this filter is faster than using overlay and pad filter to create same output.

The filter accept the following option:

inputs: Set number of input streams. Default is 2.
shortest: If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.

hue

Modify the hue and/or the saturation of the input.

It accepts the following parameters:

h: Specify the hue angle as a number of degrees. It accepts an expression, and defaults to ``0''.
s: Specify the saturation in the [-10,10] range. It accepts an expression and defaults to ``1''.
H: Specify the hue angle as a number of radians. It accepts an expression, and defaults to ``0''.
b: Specify the brightness in the [-10,10] range. It accepts an expression and defaults to ``0''.

h and H are mutually exclusive, and can't be specified at the same time.

The b, h, H and s option values are expressions containing the following constants:

n: frame count of the input frame starting from 0
pts: presentation timestamp of the input frame expressed in time base units
r: frame rate of the input video, NAN if the input frame rate is unknown
t: timestamp expressed in seconds, NAN if the input timestamp is unknown
tb: time base of the input video

Examples

*

Set the hue to 90 degrees and the saturation to 1.0:

        hue=h=90:s=1

*

Same command but expressing the hue in radians:

        hue=H=PI/2:s=1

*

Rotate hue and make the saturation swing between 0 and 2 over a period of 1 second:

        hue="H=2*PI*t: s=sin(2*PI*t)+1"

*

Apply a 3 seconds saturation fade-in effect starting at 0:

        hue="s=min(t/3\,1)"

The general fade-in expression can be written as:

        hue="s=min(0\, max((t-START)/DURATION\, 1))"

*

Apply a 3 seconds saturation fade-out effect starting at 5 seconds:

        hue="s=max(0\, min(1\, (8-t)/3))"

The general fade-out expression can be written as:

        hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))"

Commands

This filter supports the following commands:

b
s
h
H: Modify the hue and/or the saturation and/or brightness of the input video. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.

hysteresis

Grow first stream into second stream by connecting components. This makes it possible to build more robust edge masks.

This filter accepts the following options:

planes: Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream. By default value 0xf, all planes will be processed.
threshold: Set threshold which is used in filtering. If pixel component value is higher than this value filter algorithm for connecting components is activated. By default value is 0.

idet

Detect video interlacing type.

This filter tries to detect if the input frames are interlaced, progressive, top or bottom field first. It will also try to detect fields that are repeated between adjacent frames (a sign of telecine).

Single frame detection considers only immediately adjacent frames when classifying each frame. Multiple frame detection incorporates the classification history of previous frames.

The filter will log these metadata values:

single.current_frame: Detected type of current frame using single-frame detection. One of: ``tff'' (top field first), ``bff'' (bottom field first), ``progressive'', or ``undetermined''
single.tff: Cumulative number of frames detected as top field first using single-frame detection.
multiple.tff: Cumulative number of frames detected as top field first using multiple-frame detection.
single.bff: Cumulative number of frames detected as bottom field first using single-frame detection.
multiple.current_frame: Detected type of current frame using multiple-frame detection. One of: ``tff'' (top field first), ``bff'' (bottom field first), ``progressive'', or ``undetermined''
multiple.bff: Cumulative number of frames detected as bottom field first using multiple-frame detection.
single.progressive: Cumulative number of frames detected as progressive using single-frame detection.
multiple.progressive: Cumulative number of frames detected as progressive using multiple-frame detection.
single.undetermined: Cumulative number of frames that could not be classified using single-frame detection.
multiple.undetermined: Cumulative number of frames that could not be classified using multiple-frame detection.
repeated.current_frame: Which field in the current frame is repeated from the last. One of ``neither'', ``top'', or ``bottom''.
repeated.neither: Cumulative number of frames with no repeated field.
repeated.top: Cumulative number of frames with the top field repeated from the previous frame's top field.
repeated.bottom: Cumulative number of frames with the bottom field repeated from the previous frame's bottom field.

The filter accepts the following options:

intl_thres: Set interlacing threshold.
prog_thres: Set progressive threshold.
rep_thres: Threshold for repeated field detection.
half_life: Number of frames after which a given frame's contribution to the statistics is halved (i.e., it contributes only 0.5 to its classification). The default of 0 means that all frames seen are given full weight of 1.0 forever.
analyze_interlaced_flag: When this is not 0 then idet will use the specified number of frames to determine if the interlaced flag is accurate, it will not count undetermined frames. If the flag is found to be accurate it will be used without any further computations, if it is found to be inaccurate it will be cleared without any further computations. This allows inserting the idet filter as a low computational method to clean up the interlaced flag

il

Deinterleave or interleave fields.

This filter allows one to process interlaced images fields without deinterlacing them. Deinterleaving splits the input frame into 2 fields (so called half pictures). Odd lines are moved to the top half of the output image, even lines to the bottom half. You can process (filter) them independently and then re-interleave them.

The filter accepts the following options:

luma_mode, l

chroma_mode, c

alpha_mode, a

Available values for luma_mode, chroma_mode and alpha_mode are:

none: Do nothing.
deinterleave, d: Deinterleave fields, placing one above the other.
interleave, i: Interleave fields. Reverse the effect of deinterleaving.

: Default value is "none".

luma_swap, ls

chroma_swap, cs

alpha_swap, as

Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is 0.

inflate

Apply inflate effect to the video.

This filter replaces the pixel by the local(3x3) average by taking into account only values higher than the pixel.

It accepts the following options:

threshold0
threshold1
threshold2
threshold3: Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.

interlace

Simple interlacing filter from progressive contents. This interleaves upper (or lower) lines from odd frames with lower (or upper) lines from even frames, halving the frame rate and preserving image height.

           Original        Original             New Frame
           Frame 'j'      Frame 'j+1'             (tff)
          ==========      ===========       ==================
            Line 0  -------------------->    Frame 'j' Line 0
            Line 1          Line 1  ---->   Frame 'j+1' Line 1
            Line 2 --------------------->    Frame 'j' Line 2
            Line 3          Line 3  ---->   Frame 'j+1' Line 3
             ...             ...                   ...
        New Frame + 1 will be generated by Frame 'j+2' and Frame 'j+3' and so on

It accepts the following optional parameters:

scan: This determines whether the interlaced frame is taken from the even (tff - default) or odd (bff) lines of the progressive frame.
lowpass: Enable (default) or disable the vertical lowpass filter to avoid twitter interlacing and reduce moire patterns.

kerndeint

Deinterlace input video by applying Donald Graft's adaptive kernel deinterling. Work on interlaced parts of a video to produce progressive frames.

The description of the accepted parameters follows.

thresh: Set the threshold which affects the filter's tolerance when determining if a pixel line must be processed. It must be an integer in the range [0,255] and defaults to 10. A value of 0 will result in applying the process on every pixels.
map: Paint pixels exceeding the threshold value to white if set to 1. Default is 0.
order: Set the fields order. Swap fields if set to 1, leave fields alone if 0. Default is 0.
sharp: Enable additional sharpening if set to 1. Default is 0.
twoway: Enable twoway sharpening if set to 1. Default is 0.

Examples

*

Apply default values:

        kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0

*

Enable additional sharpening:

        kerndeint=sharp=1

*

Paint processed pixels in white:

        kerndeint=map=1

lenscorrection

Correct radial lens distortion

This filter can be used to correct for radial distortion as can result from the use of wide angle lenses, and thereby re-rectify the image. To find the right parameters one can use tools available for example as part of opencv or simply trial-and-error. To use opencv use the calibration sample (under samples/cpp) from the opencv sources and extract the k1 and k2 coefficients from the resulting matrix.

Note that effectively the same filter is available in the open-source tools Krita and Digikam from the KDE project.

In contrast to the vignette filter, which can also be used to compensate lens errors, this filter corrects the distortion of the image, whereas vignette corrects the brightness distribution, so you may want to use both filters together in certain cases, though you will have to take care of ordering, i.e. whether vignetting should be applied before or after lens correction.

Options

The filter accepts the following options:

cx: Relative x-coordinate of the focal point of the image, and thereby the center of the distortion. This value has a range [0,1] and is expressed as fractions of the image width.
cy: Relative y-coordinate of the focal point of the image, and thereby the center of the distortion. This value has a range [0,1] and is expressed as fractions of the image height.
k1: Coefficient of the quadratic correction term. 0.5 means no correction.
k2: Coefficient of the double quadratic correction term. 0.5 means no correction.

The formula that generates the correction is:

r_src = r_tgt * (1 + k1 * (r_tgt / r_0)^2 + k2 * (r_tgt / r_0)^4)

where r_0 is halve of the image diagonal and r_src and r_tgt are the distances from the focal point in the source and target images, respectively.

loop

Loop video frames.

The filter accepts the following options:

loop: Set the number of loops.
size: Set maximal size in number of frames.
start: Set first frame of loop.

lut3d

Apply a 3D LUT to an input video.

The filter accepts the following options:

file

Set the 3D LUT file name.

Currently supported formats:

3dl: AfterEffects
cube: Iridas
dat: DaVinci
m3d: Pandora

interp

Select interpolation mode.

Available values are:

nearest: Use values from the nearest defined point.
trilinear: Interpolate values using the 8 points defining a cube.
tetrahedral: Interpolate values using a tetrahedron.

lut, lutrgb, lutyuv

Compute a look-up table for binding each pixel component input value to an output value, and apply it to the input video.

lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video.

These filters accept the following parameters:

c0: set first pixel component expression
c1: set second pixel component expression
c2: set third pixel component expression
c3: set fourth pixel component expression, corresponds to the alpha component
r: set red component expression
g: set green component expression
b: set blue component expression
a: alpha component expression
y: set Y/luminance component expression
u: set U/Cb component expression
v: set V/Cr component expression

Each of them specifies the expression to use for computing the lookup table for the corresponding pixel component values.

The exact component associated to each of the c* options depends on the format in input.

The lut filter requires either YUV or RGB pixel formats in input, lutrgb requires RGB pixel formats in input, and lutyuv requires YUV.

The expressions can contain the following constants and functions:

w
h: The input width and height.
val: The input value for the pixel component.
clipval: The input value, clipped to the minval-maxval range.
maxval: The maximum value for the pixel component.
minval: The minimum value for the pixel component.
negval: The negated value for the pixel component value, clipped to the minval-maxval range; it corresponds to the expression ``maxval-clipval+minval''.
clip(val): The computed value in val, clipped to the minval-maxval range.
gammaval(gamma): The computed gamma correction value of the pixel component value, clipped to the minval-maxval range. It corresponds to the expression "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

All expressions default to ``val''.

Examples

*

Negate input video:

        lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
        lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"

The above is the same as:

        lutrgb="r=negval:g=negval:b=negval"
        lutyuv="y=negval:u=negval:v=negval"

*

Negate luminance:

        lutyuv=y=negval

*

Remove chroma components, turning the video into a graytone image:

        lutyuv="u=128:v=128"

*

Apply a luma burning effect:

        lutyuv="y=2*val"

*

Remove green and blue components:

        lutrgb="g=0:b=0"

*

Set a constant alpha channel value on input:

        format=rgba,lutrgb=a="maxval-minval/2"

*

Correct luminance gamma by a factor of 0.5:

        lutyuv=y=gammaval(0.5)

*

Discard least significant bits of luma:

        lutyuv=y='bitand(val, 128+64+32)'

*

Technicolor like effect:

        lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'

lut2

Compute and apply a lookup table from two video inputs.

This filter accepts the following parameters:

c0: set first pixel component expression
c1: set second pixel component expression
c2: set third pixel component expression
c3: set fourth pixel component expression, corresponds to the alpha component

Each of them specifies the expression to use for computing the lookup table for the corresponding pixel component values.

The exact component associated to each of the c* options depends on the format in inputs.

The expressions can contain the following constants:

w
h: The input width and height.
x: The first input value for the pixel component.
y: The second input value for the pixel component.
bdx: The first input video bit depth.
bdy: The second input video bit depth.

All expressions default to ``x''.

Examples

*

Highlight differences between two RGB video streams:

        lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'

*

Highlight differences between two YUV video streams:

        lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'

maskedclamp

Clamp the first input stream with the second input and third input stream.

Returns the value of first stream to be between second input stream - "undershoot" and third input stream + "overshoot".

This filter accepts the following options:

undershoot: Default value is 0.
overshoot: Default value is 0.
planes: Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream. By default value 0xf, all planes will be processed.

maskedmerge

Merge the first input stream with the second input stream using per pixel weights in the third input stream.

A value of 0 in the third stream pixel component means that pixel component from first stream is returned unchanged, while maximum value (eg. 255 for 8-bit videos) means that pixel component from second stream is returned unchanged. Intermediate values define the amount of merging between both input stream's pixel components.

This filter accepts the following options:

planes: Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream. By default value 0xf, all planes will be processed.

mcdeint

Apply motion-compensation deinterlacing.

It needs one field per frame as input and must thus be used together with yadif=1/3 or equivalent.

This filter accepts the following options:

mode

Set the deinterlacing mode.

It accepts one of the following values:

fast
medium
slow: use iterative motion estimation
extra_slow: like slow, but use multiple reference frames.

: Default value is fast.

parity

Set the picture field parity assumed for the input video. It must be one of the following values:

0, tff: assume top field first
1, bff: assume bottom field first

: Default value is bff.

qp

Set per-block quantization parameter (QP) used by the internal encoder.

Higher values should result in a smoother motion vector field but less optimal individual vectors. Default value is 1.

mergeplanes

Merge color channel components from several video streams.

The filter accepts up to 4 input streams, and merge selected input planes to the output video.

This filter accepts the following options:

mapping: Set input to output plane mapping. Default is 0.
The mappings is specified as a bitmap. It should be specified as a hexadecimal number in the form 0xAa[Bb[Cc[Dd]]]. 'Aa' describes the mapping for the first plane of the output stream. 'A' sets the number of the input stream to use (from 0 to 3), and 'a' the plane number of the corresponding input to use (from 0 to 3). The rest of the mappings is similar, 'Bb' describes the mapping for the output stream second plane, 'Cc' describes the mapping for the output stream third plane and 'Dd' describes the mapping for the output stream fourth plane.
format: Set output pixel format. Default is "yuva444p".

Examples

*

Merge three gray video streams of same width and height into single video stream:

        [a0][a1][a2]mergeplanes=0x001020:yuv444p

*

Merge 1st yuv444p stream and 2nd gray video stream into yuva444p video stream:

        [a0][a1]mergeplanes=0x00010210:yuva444p

*

Swap Y and A plane in yuva444p stream:

        format=yuva444p,mergeplanes=0x03010200:yuva444p

*

Swap U and V plane in yuv420p stream:

        format=yuv420p,mergeplanes=0x000201:yuv420p

*

Cast a rgb24 clip to yuv444p:

        format=rgb24,mergeplanes=0x000102:yuv444p

mestimate

Estimate and export motion vectors using block matching algorithms. Motion vectors are stored in frame side data to be used by other filters.

This filter accepts the following options:

method

Specify the motion estimation method. Accepts one of the following values:

esa: Exhaustive search algorithm.
tss: Three step search algorithm.
tdls: Two dimensional logarithmic search algorithm.
ntss: New three step search algorithm.
fss: Four step search algorithm.
ds: Diamond search algorithm.
hexbs: Hexagon-based search algorithm.
epzs: Enhanced predictive zonal search algorithm.
umh: Uneven multi-hexagon search algorithm.

: Default value is esa.

mb_size

Macroblock size. Default 16.

search_param

Search parameter. Default 7.

minterpolate

Convert the video to specified frame rate using motion interpolation.

This filter accepts the following options:

fps

Specify the output frame rate. This can be rational e.g. "60000/1001". Frames are dropped if fps is lower than source fps. Default 60.

mi_mode

Motion interpolation mode. Following values are accepted:

dup

Duplicate previous or next frame for interpolating new ones.

blend

Blend source frames. Interpolated frame is mean of previous and next frames.

mci

Motion compensated interpolation. Following options are effective when this mode is selected:

mc_mode

Motion compensation mode. Following values are accepted:

obmc: Overlapped block motion compensation.
aobmc: Adaptive overlapped block motion compensation. Window weighting coefficients are controlled adaptively according to the reliabilities of the neighboring motion vectors to reduce oversmoothing.

: Default mode is obmc.

me_mode

Motion estimation mode. Following values are accepted:

bidir: Bidirectional motion estimation. Motion vectors are estimated for each source frame in both forward and backward directions.
bilat: Bilateral motion estimation. Motion vectors are estimated directly for interpolated frame.

: Default mode is bilat.

me

The algorithm to be used for motion estimation. Following values are accepted:

esa: Exhaustive search algorithm.
tss: Three step search algorithm.
tdls: Two dimensional logarithmic search algorithm.
ntss: New three step search algorithm.
fss: Four step search algorithm.
ds: Diamond search algorithm.
hexbs: Hexagon-based search algorithm.
epzs: Enhanced predictive zonal search algorithm.
umh: Uneven multi-hexagon search algorithm.

: Default algorithm is epzs.

mb_size

Macroblock size. Default 16.

search_param

Motion estimation search parameter. Default 32.

vsmbc

Enable variable-size block motion compensation. Motion estimation is applied with smaller block sizes at object boundaries in order to make the them less blur. Default is 0 (disabled).

scd

Scene change detection method. Scene change leads motion vectors to be in random direction. Scene change detection replace interpolated frames by duplicate ones. May not be needed for other modes. Following values are accepted:

none: Disable scene change detection.
fdiff: Frame difference. Corresponding pixel values are compared and if it satisfies scd_threshold scene change is detected.

: Default method is fdiff.

scd_threshold

Scene change detection threshold. Default is 5.0.

mpdecimate

Drop frames that do not differ greatly from the previous frame in order to reduce frame rate.

The main use of this filter is for very-low-bitrate encoding (e.g. streaming over dialup modem), but it could in theory be used for fixing movies that were inverse-telecined incorrectly.

A description of the accepted options follows.

max

Set the maximum number of consecutive frames which can be dropped (if positive), or the minimum interval between dropped frames (if negative). If the value is 0, the frame is dropped unregarding the number of previous sequentially dropped frames.

Default value is 0.

hi

lo

frac

Set the dropping threshold values.

Values for hi and lo are for 8x8 pixel blocks and represent actual pixel value differences, so a threshold of 64 corresponds to 1 unit of difference for each pixel, or the same spread out differently over the block.

A frame is a candidate for dropping if no 8x8 blocks differ by more than a threshold of hi, and if no more than frac blocks (1 meaning the whole image) differ by more than a threshold of lo.

Default value for hi is 64*12, default value for lo is 64*5, and default value for frac is 0.33.

negate

Negate input video.

It accepts an integer in input; if non-zero it negates the alpha component (if available). The default value in input is 0.

nlmeans

Denoise frames using Non-Local Means algorithm.

Each pixel is adjusted by looking for other pixels with similar contexts. This context similarity is defined by comparing their surrounding patches of size pxp. Patches are searched in an area of rxr around the pixel.

Note that the research area defines centers for patches, which means some patches will be made of pixels outside that research area.

The filter accepts the following options.

s: Set denoising strength.
p: Set patch size.
pc: Same as p but for chroma planes.
The default value is 0 and means automatic.
r: Set research size.
rc: Same as r but for chroma planes.
The default value is 0 and means automatic.

nnedi

Deinterlace video using neural network edge directed interpolation.

This filter accepts the following options:

weights

Mandatory option, without binary file filter can not work. Currently file can be found here: https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin

deint

Set which frames to deinterlace, by default it is "all". Can be "all" or "interlaced".

field

Set mode of operation.

Can be one of the following:

af: Use frame flags, both fields.
a: Use frame flags, single field.
t: Use top field only.
b: Use bottom field only.
tf: Use both fields, top first.
bf: Use both fields, bottom first.

planes

Set which planes to process, by default filter process all frames.

nsize

Set size of local neighborhood around each pixel, used by the predictor neural network.

Can be one of the following:

s8x6
s16x6
s32x6
s48x6
s8x4
s16x4
s32x4

nns

Set the number of neurons in predicctor neural network. Can be one of the following:

n16
n32
n64
n128
n256

qual

Controls the number of different neural network predictions that are blended together to compute the final output value. Can be "fast", default or "slow".

etype

Set which set of weights to use in the predictor. Can be one of the following:

a: weights trained to minimize absolute error
s: weights trained to minimize squared error

pscrn

Controls whether or not the prescreener neural network is used to decide which pixels should be processed by the predictor neural network and which can be handled by simple cubic interpolation. The prescreener is trained to know whether cubic interpolation will be sufficient for a pixel or whether it should be predicted by the predictor nn. The computational complexity of the prescreener nn is much less than that of the predictor nn. Since most pixels can be handled by cubic interpolation, using the prescreener generally results in much faster processing. The prescreener is pretty accurate, so the difference between using it and not using it is almost always unnoticeable.

Can be one of the following:

none
original
new

: Default is "new".

fapprox

Set various debugging flags.

noformat

Force libavfilter not to use any of the specified pixel formats for the input to the next filter.

It accepts the following parameters:

pix_fmts: A '|'-separated list of pixel format names, such as apix_fmts=yuv420p|monow|rgb24".

Examples

*

Force libavfilter to use a format different from yuv420p for the input to the vflip filter:

        noformat=pix_fmts=yuv420p,vflip

*

Convert the input video to any of the formats not contained in the list:

        noformat=yuv420p|yuv444p|yuv410p

noise

Add noise on video input frame.

The filter accepts the following options:

all_seed

c0_seed

c1_seed

c2_seed

c3_seed

Set noise seed for specific pixel component or all pixel components in case of all_seed. Default value is 123457.

all_strength, alls

c0_strength, c0s

c1_strength, c1s

c2_strength, c2s

c3_strength, c3s

Set noise strength for specific pixel component or all pixel components in case all_strength. Default value is 0. Allowed range is [0, 100].

all_flags, allf

c0_flags, c0f

c1_flags, c1f

c2_flags, c2f

c3_flags, c3f

Set pixel component flags or set flags for all components if all_flags. Available values for component flags are:

a: averaged temporal noise (smoother)
p: mix random noise with a (semi)regular pattern
t: temporal noise (noise pattern changes between frames)
u: uniform noise (gaussian otherwise)

Examples

Add temporal and uniform noise to input video:

        noise=alls=20:allf=t+u

null

Pass the video source unchanged to the output.

ocr

Optical Character Recognition

This filter uses Tesseract for optical character recognition.

It accepts the following options:

datapath: Set datapath to tesseract data. Default is to use whatever was set at installation.
language: Set language, default is ``eng''.
whitelist: Set character whitelist.
blacklist: Set character blacklist.

The filter exports recognized text as the frame metadata "lavfi.ocr.text".

ocv

Apply a video transform using libopencv.

To enable this filter, install the libopencv library and headers and configure FFmpeg with "--enable-libopencv".

It accepts the following parameters:

filter_name: The name of the libopencv filter to apply.
filter_params: The parameters to pass to the libopencv filter. If not specified, the default values are assumed.

Refer to the official libopencv documentation for more precise information: <http://docs.opencv.org/master/modules/imgproc/doc/filtering.html>

Several libopencv filters are supported; see the following subsections.

dilate

Dilate an image by using a specific structuring element. It corresponds to the libopencv function "cvDilate".

It accepts the parameters: struct_el|nb_iterations.

struct_el represents a structuring element, and has the syntax: colsxrows+anchor_xxanchor_y/shape

cols and rows represent the number of columns and rows of the structuring element, anchor_x and anchor_y the anchor point, and shape the shape for the structuring element. shape must be ``rect'', ``cross'', ``ellipse'', or ``custom''.

If the value for shape is ``custom'', it must be followed by a string of the form "=filename". The file with name filename is assumed to represent a binary image, with each printable character corresponding to a bright pixel. When a custom shape is used, cols and rows are ignored, the number or columns and rows of the read file are assumed instead.

The default value for struct_el is ``3x3+0x0/rect''.

nb_iterations specifies the number of times the transform is applied to the image, and defaults to 1.

Some examples:

        # Use the default values
        ocv=dilate
        
        # Dilate using a structuring element with a 5x5 cross, iterating two times
        ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2
        
        # Read the shape from the file diamond.shape, iterating two times.
        # The file diamond.shape may contain a pattern of characters like this
        #   *
        #  ***
        # *****
        #  ***
        #   *
        # The specified columns and rows are ignored
        # but the anchor point coordinates are not
        ocv=dilate:0x0+2x2/custom=diamond.shape|2

erode

Erode an image by using a specific structuring element. It corresponds to the libopencv function "cvErode".

It accepts the parameters: struct_el:nb_iterations, with the same syntax and semantics as the dilate filter.

smooth

Smooth the input video.

The filter takes the following parameters: type|param1|param2|param3|param4.

type is the type of smooth filter to apply, and must be one of the following values: ``blur'', ``blur_no_scale'', ``median'', ``gaussian'', or ``bilateral''. The default value is ``gaussian''.

The meaning of param1, param2, param3, and param4 depend on the smooth type. param1 and param2 accept integer positive values or 0. param3 and param4 accept floating point values.

The default value for param1 is 3. The default value for the other parameters is 0.

These parameters correspond to the parameters assigned to the libopencv function "cvSmooth".

overlay

Overlay one video on top of another.

It takes two inputs and has one output. The first input is the ``main'' video on which the second input is overlaid.

It accepts the following parameters:

A description of the accepted options follows.

x

y

Set the expression for the x and y coordinates of the overlaid video on the main video. Default value is ``0'' for both expressions. In case the expression is invalid, it is set to a huge value (meaning that the overlay will not be displayed within the output visible area).

eof_action

The action to take when EOF is encountered on the secondary input; it accepts one of the following values:

repeat: Repeat the last frame (the default).
endall: End both streams.
pass: Pass the main input through.

eval

Set when the expressions for x, and y are evaluated.

It accepts the following values:

init: only evaluate expressions once during the filter initialization or when a command is processed
frame: evaluate expressions for each incoming frame

: Default value is frame.

shortest

If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.

format

Set the format for the output video.

It accepts the following values:

yuv420: force YUV420 output
yuv422: force YUV422 output
yuv444: force YUV444 output
rgb: force RGB output

: Default value is yuv420.

rgb (deprecated)

If set to 1, force the filter to accept inputs in the RGB color space. Default value is 0. This option is deprecated, use format instead.

repeatlast

If set to 1, force the filter to draw the last overlay frame over the main input until the end of the stream. A value of 0 disables this behavior. Default value is 1.

The x, and y expressions can contain the following parameters.

main_w, W
main_h, H: The main input width and height.
overlay_w, w
overlay_h, h: The overlay input width and height.
x
y: The computed values for x and y. They are evaluated for each new frame.
hsub
vsub: horizontal and vertical chroma subsample values of the output format. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.
n: the number of input frame, starting from 0
pos: the position in the file of the input frame, NAN if unknown
t: The timestamp, expressed in seconds. It's NAN if the input timestamp is unknown.

Note that the n, pos, t variables are available only when evaluation is done per frame, and will evaluate to NAN when eval is set to init.

Be aware that frames are taken from each input video in timestamp order, hence, if their initial timestamps differ, it is a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp, as the example for the movie filter does.

You can chain together more overlays but you should test the efficiency of such approach.

Commands

This filter supports the following commands:

x
y: Modify the x and y of the overlay input. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.

Examples

*

Draw the overlay at 10 pixels from the bottom right corner of the main video:

        overlay=main_w-overlay_w-10:main_h-overlay_h-10

Using named options the example above becomes:

        overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10

*

Insert a transparent PNG logo in the bottom left corner of the input, using the ffmpeg tool with the "-filter_complex" option:

        ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output

*

Insert 2 different transparent PNG logos (second logo on bottom right corner) using the ffmpeg tool:

        ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output

*

Add a transparent color layer on top of the main video; "WxH" must specify the size of the main input to the overlay filter:

        color=color=red@.3:size=WxH [over]; [in][over] overlay [out]

*

Play an original video and a filtered version (here with the deshake filter) side by side using the ffplay tool:

        ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'

The above command is the same as:

        ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'

*

Make a sliding overlay appearing from the left to the right top part of the screen starting since time 2:

        overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0

*

Compose output by putting two input videos side to side:

        ffmpeg -i left.avi -i right.avi -filter_complex "
        nullsrc=size=200x100 [background];
        [0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
        [1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
        [background][left]       overlay=shortest=1       [background+left];
        [background+left][right] overlay=shortest=1:x=100 [left+right]
        "

*

Mask 10-20 seconds of a video by applying the delogo filter to a section

        ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
        -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
        masked.avi

*

Chain several overlays in cascade:

        nullsrc=s=200x200 [bg];
        testsrc=s=100x100, split=4 [in0][in1][in2][in3];
        [in0] lutrgb=r=0, [bg]   overlay=0:0     [mid0];
        [in1] lutrgb=g=0, [mid0] overlay=100:0   [mid1];
        [in2] lutrgb=b=0, [mid1] overlay=0:100   [mid2];
        [in3] null,       [mid2] overlay=100:100 [out0]

owdenoise

Apply Overcomplete Wavelet denoiser.

The filter accepts the following options:

depth

Set depth.

Larger depth values will denoise lower frequency components more, but slow down filtering.

Must be an int in the range 8-16, default is 8.

luma_strength, ls

Set luma strength.

Must be a double value in the range 0-1000, default is 1.0.

chroma_strength, cs

Set chroma strength.

Must be a double value in the range 0-1000, default is 1.0.

pad

Add paddings to the input image, and place the original input at the provided x, y coordinates.

It accepts the following parameters:

width, w

height, h

Specify an expression for the size of the output image with the paddings added. If the value for width or height is 0, the corresponding input size is used for the output.

The width expression can reference the value set by the height expression, and vice versa.

The default value of width and height is 0.

x

y

Specify the offsets to place the input image at within the padded area, with respect to the top/left border of the output image.

The x expression can reference the value set by the y expression, and vice versa.

The default value of x and y is 0.

color

Specify the color of the padded area. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of color is ``black''.

The value for the width, height, x, and y options are expressions containing the following constants:

in_w
in_h: The input video width and height.
iw
ih: These are the same as in_w and in_h.
out_w
out_h: The output width and height (the size of the padded area), as specified by the width and height expressions.
ow
oh: These are the same as out_w and out_h.
x
y: The x and y offsets as specified by the x and y expressions, or NAN if not yet specified.
a: same as iw / ih
sar: input sample aspect ratio
dar: input display aspect ratio, it is the same as (iw / ih) * sar
hsub
vsub: The horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

Examples

*

Add paddings with the color ``violet'' to the input video. The output video size is 640x480, and the top-left corner of the input video is placed at column 0, row 40

        pad=640:480:0:40:violet

The example above is equivalent to the following command:

        pad=width=640:height=480:x=0:y=40:color=violet

*

Pad the input to get an output with dimensions increased by 3/2, and put the input video at the center of the padded area:

        pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"

*

Pad the input to get a squared output with size equal to the maximum value between the input width and height, and put the input video at the center of the padded area:

        pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"

*

Pad the input to get a final w/h ratio of 16:9:

        pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"

*

In case of anamorphic video, in order to set the output display aspect correctly, it is necessary to use sar in the expression, according to the relation:

        (ih * X / ih) * sar = output_dar
        X = output_dar / sar

Thus the previous example needs to be modified to:

        pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"

*

Double the output size and put the input video in the bottom-right corner of the output padded area:

        pad="2*iw:2*ih:ow-iw:oh-ih"

palettegen

Generate one palette for a whole video stream.

It accepts the following options:

max_colors

Set the maximum number of colors to quantize in the palette. Note: the palette will still contain 256 colors; the unused palette entries will be black.

reserve_transparent

Create a palette of 255 colors maximum and reserve the last one for transparency. Reserving the transparency color is useful for GIF optimization. If not set, the maximum of colors in the palette will be 256. You probably want to disable this option for a standalone image. Set by default.

stats_mode

Set statistics mode.

It accepts the following values:

full: Compute full frame histograms.
diff: Compute histograms only for the part that differs from previous frame. This might be relevant to give more importance to the moving part of your input if the background is static.
single: Compute new histogram for each frame.

: Default value is full.

The filter also exports the frame metadata "lavfi.color_quant_ratio" ("nb_color_in / nb_color_out") which you can use to evaluate the degree of color quantization of the palette. This information is also visible at info logging level.

Examples

*

Generate a representative palette of a given video using ffmpeg:

        ffmpeg -i input.mkv -vf palettegen palette.png

paletteuse

Use a palette to downsample an input video stream.

The filter takes two inputs: one video stream and a palette. The palette must be a 256 pixels image.

It accepts the following options:

dither

Select dithering mode. Available algorithms are:

bayer: Ordered 8x8 bayer dithering (deterministic)
heckbert: Dithering as defined by Paul Heckbert in 1982 (simple error diffusion). Note: this dithering is sometimes considered ``wrong'' and is included as a reference.
floyd_steinberg: Floyd and Steingberg dithering (error diffusion)
sierra2: Frankie Sierra dithering v2 (error diffusion)
sierra2_4a: Frankie Sierra dithering v2 ``Lite'' (error diffusion)

: Default is sierra2_4a.

bayer_scale

When bayer dithering is selected, this option defines the scale of the pattern (how much the crosshatch pattern is visible). A low value means more visible pattern for less banding, and higher value means less visible pattern at the cost of more banding.

The option must be an integer value in the range [0,5]. Default is 2.

diff_mode

If set, define the zone to process

rectangle: Only the changing rectangle will be reprocessed. This is similar to GIF cropping/offsetting compression mechanism. This option can be useful for speed if only a part of the image is changing, and has use cases such as limiting the scope of the error diffusal dither to the rectangle that bounds the moving scene (it leads to more deterministic output if the scene doesn't change much, and as a result less moving noise and better GIF compression).

: Default is none.

new

Take new palette for each output frame.

Examples

*

Use a palette (generated for example with palettegen) to encode a GIF using ffmpeg:

        ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif

perspective

Correct perspective of video not recorded perpendicular to the screen.

A description of the accepted parameters follows.

x0

y0

x1

y1

x2

y2

x3

y3

Set coordinates expression for top left, top right, bottom left and bottom right corners. Default values are "0:0:W:0:0:H:W:H" with which perspective will remain unchanged. If the "sense" option is set to "source", then the specified points will be sent to the corners of the destination. If the "sense" option is set to "destination", then the corners of the source will be sent to the specified coordinates.

The expressions can use the following variables:

W
H: the width and height of video frame.
in: Input frame count.
on: Output frame count.

interpolation

Set interpolation for perspective correction.

It accepts the following values:

linear
cubic

: Default value is linear.

sense

Set interpretation of coordinate options.

It accepts the following values:

0, source: Send point in the source specified by the given coordinates to the corners of the destination.
1, destination: Send the corners of the source to the point in the destination specified by the given coordinates.
Default value is source.

eval

Set when the expressions for coordinates x0,y0,...x3,y3 are evaluated.

It accepts the following values:

init: only evaluate expressions once during the filter initialization or when a command is processed
frame: evaluate expressions for each incoming frame

: Default value is init.

phase

Delay interlaced video by one field time so that the field order changes.

The intended use is to fix PAL movies that have been captured with the opposite field order to the film-to-video transfer.

A description of the accepted parameters follows.

mode

Set phase mode.

It accepts the following values:

t: Capture field order top-first, transfer bottom-first. Filter will delay the bottom field.
b: Capture field order bottom-first, transfer top-first. Filter will delay the top field.
p: Capture and transfer with the same field order. This mode only exists for the documentation of the other options to refer to, but if you actually select it, the filter will faithfully do nothing.
a: Capture field order determined automatically by field flags, transfer opposite. Filter selects among t and b modes on a frame by frame basis using field flags. If no field information is available, then this works just like u.
u: Capture unknown or varying, transfer opposite. Filter selects among t and b on a frame by frame basis by analyzing the images and selecting the alternative that produces best match between the fields.
T: Capture top-first, transfer unknown or varying. Filter selects among t and p using image analysis.
B: Capture bottom-first, transfer unknown or varying. Filter selects among b and p using image analysis.
A: Capture determined by field flags, transfer unknown or varying. Filter selects among t, b and p using field flags and image analysis. If no field information is available, then this works just like U. This is the default mode.
U: Both capture and transfer unknown or varying. Filter selects among t, b and p using image analysis only.

pixdesctest

Pixel format descriptor test filter, mainly useful for internal testing. The output video should be equal to the input video.

For example:

        format=monow, pixdesctest

can be used to test the monowhite pixel format descriptor definition.

pp

Enable the specified chain of postprocessing subfilters using libpostproc. This library should be automatically selected with a GPL build ("--enable-gpl"). Subfilters must be separated by '/' and can be disabled by prepending a '-'. Each subfilter and some options have a short and a long name that can be used interchangeably, i.e. dr/dering are the same.

The filters accept the following options:

subfilters: Set postprocessing subfilters string.

All subfilters share common options to determine their scope:

a/autoq: Honor the quality commands for this subfilter.
c/chrom: Do chrominance filtering, too (default).
y/nochrom: Do luminance filtering only (no chrominance).
n/noluma: Do chrominance filtering only (no luminance).

These options can be appended after the subfilter name, separated by a '|'.

Available subfilters are:

hb/hdeblock[|difference[|flatness]]

Horizontal deblocking filter

difference: Difference factor where higher values mean more deblocking (default: 32).
flatness: Flatness threshold where lower values mean more deblocking (default: 39).

vb/vdeblock[|difference[|flatness]]

Vertical deblocking filter

difference: Difference factor where higher values mean more deblocking (default: 32).
flatness: Flatness threshold where lower values mean more deblocking (default: 39).

ha/hadeblock[|difference[|flatness]]

Accurate horizontal deblocking filter

difference: Difference factor where higher values mean more deblocking (default: 32).
flatness: Flatness threshold where lower values mean more deblocking (default: 39).

va/vadeblock[|difference[|flatness]]

Accurate vertical deblocking filter

difference: Difference factor where higher values mean more deblocking (default: 32).
flatness: Flatness threshold where lower values mean more deblocking (default: 39).

The horizontal and vertical deblocking filters share the difference and flatness values so you cannot set different horizontal and vertical thresholds.

h1/x1hdeblock

Experimental horizontal deblocking filter

v1/x1vdeblock

Experimental vertical deblocking filter

dr/dering

Deringing filter

tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise reducer

threshold1: larger -> stronger filtering
threshold2: larger -> stronger filtering
threshold3: larger -> stronger filtering

al/autolevels[:f/fullyrange], automatic brightness / contrast correction

f/fullyrange: Stretch luminance to "0-255".

lb/linblenddeint

Linear blend deinterlacing filter that deinterlaces the given block by filtering all lines with a "(1 2 1)" filter.

li/linipoldeint

Linear interpolating deinterlacing filter that deinterlaces the given block by linearly interpolating every second line.

ci/cubicipoldeint

Cubic interpolating deinterlacing filter deinterlaces the given block by cubically interpolating every second line.

md/mediandeint

Median deinterlacing filter that deinterlaces the given block by applying a median filter to every second line.

fd/ffmpegdeint

FFmpeg deinterlacing filter that deinterlaces the given block by filtering every second line with a "(-1 4 2 4 -1)" filter.

l5/lowpass5

Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given block by filtering all lines with a "(-1 2 6 2 -1)" filter.

fq/forceQuant[|quantizer]

Overrides the quantizer table from the input with the constant quantizer you specify.

quantizer: Quantizer to use

de/default

Default pp filter combination ("hb|a,vb|a,dr|a")

fa/fast

Fast pp filter combination ("h1|a,v1|a,dr|a")

ac

High quality pp filter combination ("ha|a|128|7,va|a,dr|a")

Examples

*

Apply horizontal and vertical deblocking, deringing and automatic brightness/contrast:

        pp=hb/vb/dr/al

*

Apply default filters without brightness/contrast correction:

        pp=de/-al

*

Apply default filters and temporal denoiser:

        pp=default/tmpnoise|1|2|3

*

Apply deblocking on luminance only, and switch vertical deblocking on or off automatically depending on available CPU time:

        pp=hb|y/vb|a

pp7

Apply Postprocessing filter 7. It is variant of the spp filter, similar to spp = 6 with 7 point DCT, where only the center sample is used after IDCT.

The filter accepts the following options:

qp

Force a constant quantization parameter. It accepts an integer in range 0 to 63. If not set, the filter will use the QP from the video stream (if available).

mode

Set thresholding mode. Available modes are:

hard: Set hard thresholding.
soft: Set soft thresholding (better de-ringing effect, but likely blurrier).
medium: Set medium thresholding (good results, default).

prewitt

Apply prewitt operator to input video stream.

The filter accepts the following option:

planes: Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
scale: Set value which will be multiplied with filtered result.
delta: Set value which will be added to filtered result.

psnr

Obtain the average, maximum and minimum PSNR (Peak Signal to Noise Ratio) between two input videos.

This filter takes in input two input videos, the first input is considered the ``main'' source and is passed unchanged to the output. The second input is used as a ``reference'' video for computing the PSNR.

Both video inputs must have the same resolution and pixel format for this filter to work correctly. Also it assumes that both inputs have the same number of frames, which are compared one by one.

The obtained average PSNR is printed through the logging system.

The filter stores the accumulated MSE (mean squared error) of each frame, and at the end of the processing it is averaged across all frames equally, and the following formula is applied to obtain the PSNR:

        PSNR = 10*log10(MAX^2/MSE)

Where MAX is the average of the maximum values of each component of the image.

The description of the accepted parameters follows.

stats_file, f: If specified the filter will use the named file to save the PSNR of each individual frame. When filename equals ``-'' the data is sent to standard output.
stats_version: Specifies which version of the stats file format to use. Details of each format are written below. Default value is 1.
stats_add_max: Determines whether the max value is output to the stats log. Default value is 0. Requires stats_version >= 2. If this is set and stats_version < 2, the filter will return an error.

The file printed if stats_file is selected, contains a sequence of key/value pairs of the form key:value for each compared couple of frames.

If a stats_version greater than 1 is specified, a header line precedes the list of per-frame-pair stats, with key value pairs following the frame format with the following parameters:

psnr_log_version: The version of the log file format. Will match stats_version.
fields: A comma separated list of the per-frame-pair parameters included in the log.

A description of each shown per-frame-pair parameter follows:

n: sequential number of the input frame, starting from 1
mse_avg: Mean Square Error pixel-by-pixel average difference of the compared frames, averaged over all the image components.
mse_y, mse_u, mse_v, mse_r, mse_g, mse_g, mse_a: Mean Square Error pixel-by-pixel average difference of the compared frames for the component specified by the suffix.
psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a: Peak Signal to Noise ratio of the compared frames for the component specified by the suffix.
max_avg, max_y, max_u, max_v: Maximum allowed value for each channel, and average over all channels.

For example:

        movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
        [main][ref] psnr="stats_file=stats.log" [out]

On this example the input file being processed is compared with the reference file ref_movie.mpg. The PSNR of each individual frame is stored in stats.log.

pullup

Pulldown reversal (inverse telecine) filter, capable of handling mixed hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps progressive content.

The pullup filter is designed to take advantage of future context in making its decisions. This filter is stateless in the sense that it does not lock onto a pattern to follow, but it instead looks forward to the following fields in order to identify matches and rebuild progressive frames.

To produce content with an even framerate, insert the fps filter after pullup, use "fps=24000/1001" if the input frame rate is 29.97fps, "fps=24" for 30fps and the (rare) telecined 25fps input.

The filter accepts the following options:

jl

jr

jt

jb

These options set the amount of ``junk'' to ignore at the left, right, top, and bottom of the image, respectively. Left and right are in units of 8 pixels, while top and bottom are in units of 2 lines. The default is 8 pixels on each side.

sb

Set the strict breaks. Setting this option to 1 will reduce the chances of filter generating an occasional mismatched frame, but it may also cause an excessive number of frames to be dropped during high motion sequences. Conversely, setting it to -1 will make filter match fields more easily. This may help processing of video where there is slight blurring between the fields, but may also cause there to be interlaced frames in the output. Default value is 0.

mp

Set the metric plane to use. It accepts the following values:

l: Use luma plane.
u: Use chroma blue plane.
v: Use chroma red plane.

: This option may be set to use chroma plane instead of the default luma plane for doing filter's computations. This may improve accuracy on very clean source material, but more likely will decrease accuracy, especially if there is chroma noise (rainbow effect) or any grayscale video. The main purpose of setting mp to a chroma plane is to reduce CPU load and make pullup usable in realtime on slow machines.

For best results (without duplicated frames in the output file) it is necessary to change the output frame rate. For example, to inverse telecine NTSC input:

        ffmpeg -i input -vf pullup -r 24000/1001 ...

qp

Change video quantization parameters (QP).

The filter accepts the following option:

qp: Set expression for quantization parameter.

The expression is evaluated through the eval API and can contain, among others, the following constants:

known: 1 if index is not 129, 0 otherwise.
qp: Sequentional index starting from -129 to 128.

Examples

*

Some equation like:

        qp=2+2*sin(PI*qp)

random

Flush video frames from internal cache of frames into a random order. No frame is discarded. Inspired by frei0r nervous filter.

frames: Set size in number of frames of internal cache, in range from 2 to 512. Default is 30.
seed: Set seed for random number generator, must be an integer included between 0 and "UINT32_MAX". If not specified, or if explicitly set to less than 0, the filter will try to use a good random seed on a best effort basis.

readvitc

Read vertical interval timecode (VITC) information from the top lines of a video frame.

The filter adds frame metadata key "lavfi.readvitc.tc_str" with the timecode value, if a valid timecode has been detected. Further metadata key "lavfi.readvitc.found" is set to 0/1 depending on whether timecode data has been found or not.

This filter accepts the following options:

scan_max: Set the maximum number of lines to scan for VITC data. If the value is set to "-1" the full video frame is scanned. Default is 45.
thr_b: Set the luma threshold for black. Accepts float numbers in the range [0.0,1.0], default value is 0.2. The value must be equal or less than "thr_w".
thr_w: Set the luma threshold for white. Accepts float numbers in the range [0.0,1.0], default value is 0.6. The value must be equal or greater than "thr_b".

Examples

*

Detect and draw VITC data onto the video frame; if no valid VITC is detected, draw "--:--:--:--" as a placeholder:

        ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'

remap

Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.

Destination pixel at position (X, Y) will be picked from source (x, y) position where x = Xmap(X, Y) and y = Ymap(X, Y). If mapping values are out of range, zero value for pixel will be used for destination pixel.

Xmap and Ymap input video streams must be of same dimensions. Output video stream will have Xmap/Ymap video stream dimensions. Xmap and Ymap input video streams are 16bit depth, single channel.

removegrain

The removegrain filter is a spatial denoiser for progressive video.

m0: Set mode for the first plane.
m1: Set mode for the second plane.
m2: Set mode for the third plane.
m3: Set mode for the fourth plane.

Range of mode is from 0 to 24. Description of each mode follows:

0: Leave input plane unchanged. Default.
1: Clips the pixel with the minimum and maximum of the 8 neighbour pixels.
2: Clips the pixel with the second minimum and maximum of the 8 neighbour pixels.
3: Clips the pixel with the third minimum and maximum of the 8 neighbour pixels.
4: Clips the pixel with the fourth minimum and maximum of the 8 neighbour pixels. This is equivalent to a median filter.
5: Line-sensitive clipping giving the minimal change.
6: Line-sensitive clipping, intermediate.
7: Line-sensitive clipping, intermediate.
8: Line-sensitive clipping, intermediate.
9: Line-sensitive clipping on a line where the neighbours pixels are the closest.
10: Replaces the target pixel with the closest neighbour.
11: [1 2 1] horizontal and vertical kernel blur.
12: Same as mode 11.
13: Bob mode, interpolates top field from the line where the neighbours pixels are the closest.
14: Bob mode, interpolates bottom field from the line where the neighbours pixels are the closest.
15: Bob mode, interpolates top field. Same as 13 but with a more complicated interpolation formula.
16: Bob mode, interpolates bottom field. Same as 14 but with a more complicated interpolation formula.
17: Clips the pixel with the minimum and maximum of respectively the maximum and minimum of each pair of opposite neighbour pixels.
18: Line-sensitive clipping using opposite neighbours whose greatest distance from the current pixel is minimal.
19: Replaces the pixel with the average of its 8 neighbours.
20: Averages the 9 pixels ([1 1 1] horizontal and vertical blur).
21: Clips pixels using the averages of opposite neighbour.
22: Same as mode 21 but simpler and faster.
23: Small edge and halo removal, but reputed useless.
24: Similar as 23.

removelogo

Suppress a TV station logo, using an image file to determine which pixels comprise the logo. It works by filling in the pixels that comprise the logo with neighboring pixels.

The filter accepts the following options:

filename, f: Set the filter bitmap file, which can be any image format supported by libavformat. The width and height of the image file must match those of the video stream being processed.

Pixels in the provided bitmap image with a value of zero are not considered part of the logo, non-zero pixels are considered part of the logo. If you use white (255) for the logo and black (0) for the rest, you will be safe. For making the filter bitmap, it is recommended to take a screen capture of a black frame with the logo visible, and then using a threshold filter followed by the erode filter once or twice.

If needed, little splotches can be fixed manually. Remember that if logo pixels are not covered, the filter quality will be much reduced. Marking too many pixels as part of the logo does not hurt as much, but it will increase the amount of blurring needed to cover over the image and will destroy more information than necessary, and extra pixels will slow things down on a large logo.

repeatfields

This filter uses the repeat_field flag from the Video ES headers and hard repeats fields based on its value.

reverse

Reverse a video clip.

Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.

Examples

*

Take the first 5 seconds of a clip, and reverse it.

        trim=end=5,reverse

rotate

Rotate video by an arbitrary angle expressed in radians.

The filter accepts the following options:

A description of the optional parameters follows.

angle, a: Set an expression for the angle by which to rotate the input video clockwise, expressed as a number of radians. A negative value will result in a counter-clockwise rotation. By default it is set to ``0''.
This expression is evaluated for each frame.
out_w, ow: Set the output width expression, default value is ``iw''. This expression is evaluated just once during configuration.
out_h, oh: Set the output height expression, default value is ``ih''. This expression is evaluated just once during configuration.
bilinear: Enable bilinear interpolation if set to 1, a value of 0 disables it. Default value is 1.
fillcolor, c: Set the color used to fill the output area not covered by the rotated image. For the general syntax of this option, check the ``Color'' section in the ffmpeg-utils manual. If the special value ``none'' is selected then no background is printed (useful for example if the background is never shown).
Default value is ``black''.

The expressions for the angle and the output size can contain the following constants and functions:

n: sequential number of the input frame, starting from 0. It is always NAN before the first frame is filtered.
t: time in seconds of the input frame, it is set to 0 when the filter is configured. It is always NAN before the first frame is filtered.
hsub
vsub: horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.
in_w, iw
in_h, ih: the input video width and height
out_w, ow
out_h, oh: the output width and height, that is the size of the padded area as specified by the width and height expressions
rotw(a)
roth(a): the minimal width/height required for completely containing the input video rotated by a radians.
These are only available when computing the out_w and out_h expressions.

Examples

*

Rotate the input by PI/6 radians clockwise:

        rotate=PI/6

*

Rotate the input by PI/6 radians counter-clockwise:

        rotate=-PI/6

*

Rotate the input by 45 degrees clockwise:

        rotate=45*PI/180

*

Apply a constant rotation with period T, starting from an angle of PI/3:

        rotate=PI/3+2*PI*t/T

*

Make the input video rotation oscillating with a period of T seconds and an amplitude of A radians:

        rotate=A*sin(2*PI/T*t)

*

Rotate the video, output size is chosen so that the whole rotating input video is always completely contained in the output:

        rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'

*

Rotate the video, reduce the output size so that no background is ever shown:

        rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none

Commands

The filter supports the following commands:

a, angle: Set the angle expression. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.

sab

Apply Shape Adaptive Blur.

The filter accepts the following options:

luma_radius, lr: Set luma blur filter strength, must be a value in range 0.1-4.0, default value is 1.0. A greater value will result in a more blurred image, and in slower processing.
luma_pre_filter_radius, lpfr: Set luma pre-filter radius, must be a value in the 0.1-2.0 range, default value is 1.0.
luma_strength, ls: Set luma maximum difference between pixels to still be considered, must be a value in the 0.1-100.0 range, default value is 1.0.
chroma_radius, cr: Set chroma blur filter strength, must be a value in range -0.9-4.0. A greater value will result in a more blurred image, and in slower processing.
chroma_pre_filter_radius, cpfr: Set chroma pre-filter radius, must be a value in the -0.9-2.0 range.
chroma_strength, cs: Set chroma maximum difference between pixels to still be considered, must be a value in the -0.9-100.0 range.

Each chroma option value, if not explicitly specified, is set to the corresponding luma option value.

scale

Scale (resize) the input video, using the libswscale library.

The scale filter forces the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio.

If the input image format is different from the format requested by the next filter, the scale filter will convert the input to the requested format.

Options

The filter accepts the following options, or any of the options supported by the libswscale scaler.

See the ffmpeg-scaler manual for the complete list of scaler options.

width, w

height, h

Set the output video dimension expression. Default value is the input dimension.

If the value is 0, the input width is used for the output.

If one of the values is -1, the scale filter will use a value that maintains the aspect ratio of the input image, calculated from the other specified dimension. If both of them are -1, the input size is used

If one of the values is -n with n > 1, the scale filter will also use a value that maintains the aspect ratio of the input image, calculated from the other specified dimension. After that it will, however, make sure that the calculated dimension is divisible by n and adjust the value if necessary.

See below for the list of accepted constants for use in the dimension expression.

eval

Specify when to evaluate width and height expression. It accepts the following values:

init: Only evaluate expressions once during the filter initialization or when a command is processed.
frame: Evaluate expressions for each incoming frame.

: Default value is init.

interl

Set the interlacing mode. It accepts the following values:

1: Force interlaced aware scaling.
0: Do not apply interlaced scaling.
-1: Select interlaced aware scaling depending on whether the source frames are flagged as interlaced or not.

: Default value is 0.

flags

Set libswscale scaling flags. See the ffmpeg-scaler manual for the complete list of values. If not explicitly specified the filter applies the default flags.

param0, param1

Set libswscale input parameters for scaling algorithms that need them. See the ffmpeg-scaler manual for the complete documentation. If not explicitly specified the filter applies empty parameters.

size, s

Set the video size. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual.

in_color_matrix

out_color_matrix

Set in/output YCbCr color space type.

This allows the autodetected value to be overridden as well as allows forcing a specific value used for the output and encoder.

If not specified, the color space type depends on the pixel format.

Possible values:

auto

Choose automatically.

bt709

Format conforming to International Telecommunication Union (ITU) Recommendation BT.709.

fcc

Set color space conforming to the United States Federal Communications Commission (FCC) Code of Federal Regulations (CFR) Title 47 (2003) 73.682 (a).

bt601

Set color space conforming to:

*: ITU Radiocommunication Sector (ITU-R) Recommendation BT.601
*: ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G
*: Society of Motion Picture and Television Engineers (SMPTE) ST 170:2004

smpte240m

Set color space conforming to SMPTE ST 240:1999.

in_range

out_range

Set in/output YCbCr sample range.

This allows the autodetected value to be overridden as well as allows forcing a specific value used for the output and encoder. If not specified, the range depends on the pixel format. Possible values:

auto: Choose automatically.
jpeg/full/pc: Set full range (0-255 in case of 8-bit luma).
mpeg/tv: Set ``MPEG'' range (16-235 in case of 8-bit luma).

force_original_aspect_ratio

Enable decreasing or increasing output video width or height if necessary to keep the original aspect ratio. Possible values:

disable: Scale the video as specified and disable this feature.
decrease: The output video dimensions will automatically be decreased if needed.
increase: The output video dimensions will automatically be increased if needed.

One useful instance of this option is that when you know a specific device's maximum allowed resolution, you can use this to limit the output video to that, while retaining the aspect ratio. For example, device A allows 1280x720 playback, and your video is 1920x800. Using this option (set it to decrease) and specifying 1280x720 to the command line makes the output 1280x533.

Please note that this is a different thing than specifying -1 for w or h, you still need to specify the output resolution for this option to work.

The values of the w and h options are expressions containing the following constants:

in_w
in_h: The input width and height
iw
ih: These are the same as in_w and in_h.
out_w
out_h: The output (scaled) width and height
ow
oh: These are the same as out_w and out_h
a: The same as iw / ih
sar: input sample aspect ratio
dar: The input display aspect ratio. Calculated from "(iw / ih) * sar".
hsub
vsub: horizontal and vertical input chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.
ohsub
ovsub: horizontal and vertical output chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

Examples

*

Scale the input video to a size of 200x100

        scale=w=200:h=100

This is equivalent to:

        scale=200:100

or:

        scale=200x100

*

Specify a size abbreviation for the output size:

        scale=qcif

which can also be written as:

        scale=size=qcif

*

Scale the input to 2x:

        scale=w=2*iw:h=2*ih

*

The above is the same as:

        scale=2*in_w:2*in_h

*

Scale the input to 2x with forced interlaced scaling:

        scale=2*iw:2*ih:interl=1

*

Scale the input to half size:

        scale=w=iw/2:h=ih/2

*

Increase the width, and set the height to the same size:

        scale=3/2*iw:ow

*

Seek Greek harmony:

        scale=iw:1/PHI*iw
        scale=ih*PHI:ih

*

Increase the height, and set the width to 3/2 of the height:

        scale=w=3/2*oh:h=3/5*ih

*

Increase the size, making the size a multiple of the chroma subsample values:

        scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"

*

Increase the width to a maximum of 500 pixels, keeping the same aspect ratio as the input:

        scale=w='min(500\, iw*3/2):h=-1'

Commands

This filter supports the following commands:

width, w
height, h: Set the output video dimension expression. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.

scale_npp

Use the NVIDIA Performance Primitives (libnpp) to perform scaling and/or pixel format conversion on CUDA video frames. Setting the output width and height works in the same way as for the scale filter.

The following additional options are accepted:

format

The pixel format of the output CUDA frames. If set to the string ``same'' (the default), the input format will be kept. Note that automatic format negotiation and conversion is not yet supported for hardware frames

interp_algo

The interpolation algorithm used for resizing. One of the following:

nn: Nearest neighbour.
linear
cubic
cubic2p_bspline: 2-parameter cubic (B=1, C=0)
cubic2p_catmullrom: 2-parameter cubic (B=0, C=1/2)
cubic2p_b05c03: 2-parameter cubic (B=1/2, C=3/10)
super: Supersampling
lanczos

scale2ref

Scale (resize) the input video, based on a reference video.

See the scale filter for available options, scale2ref supports the same but uses the reference video instead of the main input as basis.

Examples

*

Scale a subtitle stream to match the main video in size before overlaying

        'scale2ref[b][a];[a][b]overlay'

selectivecolor

Adjust cyan, magenta, yellow and black (CMYK) to certain ranges of colors (such as ``reds'', ``yellows'', ``greens'', ``cyans'', ...). The adjustment range is defined by the ``purity'' of the color (that is, how saturated it already is).

This filter is similar to the Adobe Photoshop Selective Color tool.

The filter accepts the following options:

correction_method

Select color correction method.

Available values are:

absolute: Specified adjustments are applied ``as-is'' (added/subtracted to original pixel component value).
relative: Specified adjustments are relative to the original component value.

: Default is "absolute".

reds

Adjustments for red pixels (pixels where the red component is the maximum)

yellows

Adjustments for yellow pixels (pixels where the blue component is the minimum)

greens

Adjustments for green pixels (pixels where the green component is the maximum)

cyans

Adjustments for cyan pixels (pixels where the red component is the minimum)

blues

Adjustments for blue pixels (pixels where the blue component is the maximum)

magentas

Adjustments for magenta pixels (pixels where the green component is the minimum)

whites

Adjustments for white pixels (pixels where all components are greater than 128)

neutrals

Adjustments for all pixels except pure black and pure white

blacks

Adjustments for black pixels (pixels where all components are lesser than 128)

psfile

Specify a Photoshop selective color file (".asv") to import the settings from.

All the adjustment settings (reds, yellows, ...) accept up to 4 space separated floating point adjustment values in the [-1,1] range, respectively to adjust the amount of cyan, magenta, yellow and black for the pixels of its range.

Examples

*

Increase cyan by 50% and reduce yellow by 33% in every green areas, and increase magenta by 27% in blue areas:

        selectivecolor=greens=.5 0 -.33 0:blues=0 .27

*

Use a Photoshop selective color preset:

        selectivecolor=psfile=MySelectiveColorPresets/Misty.asv

separatefields

The "separatefields" takes a frame-based video input and splits each frame into its components fields, producing a new half height clip with twice the frame rate and twice the frame count.

This filter use field-dominance information in frame to decide which of each pair of fields to place first in the output. If it gets it wrong use setfield filter before "separatefields" filter.

setdar, setsar

The "setdar" filter sets the Display Aspect Ratio for the filter output video.

This is done by changing the specified Sample (aka Pixel) Aspect Ratio, according to the following equation:

        <DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR>

Keep in mind that the "setdar" filter does not modify the pixel dimensions of the video frame. Also, the display aspect ratio set by this filter may be changed by later filters in the filterchain, e.g. in case of scaling or if another ``setdar'' or a ``setsar'' filter is applied.

The "setsar" filter sets the Sample (aka Pixel) Aspect Ratio for the filter output video.

Note that as a consequence of the application of this filter, the output display aspect ratio will change according to the equation above.

Keep in mind that the sample aspect ratio set by the "setsar" filter may be changed by later filters in the filterchain, e.g. if another ``setsar'' or a ``setdar'' filter is applied.

It accepts the following parameters:

r, ratio, dar (setdar only), sar (setsar only): Set the aspect ratio used by the filter.
The parameter can be a floating point number string, an expression, or a string of the form num:den, where num and den are the numerator and denominator of the aspect ratio. If the parameter is not specified, it is assumed the value ``0''. In case the form "num:den" is used, the ":" character should be escaped.
max: Set the maximum integer value to use for expressing numerator and denominator when reducing the expressed aspect ratio to a rational. Default value is 100.

The parameter sar is an expression containing the following constants:

E, PI, PHI: These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).
w, h: The input width and height.
a: These are the same as w / h.
sar: The input sample aspect ratio.
dar: The input display aspect ratio. It is the same as (w / h) * sar.
hsub, vsub: Horizontal and vertical chroma subsample values. For example, for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

Examples

*

To change the display aspect ratio to 16:9, specify one of the following:

        setdar=dar=1.77777
        setdar=dar=16/9

*

To change the sample aspect ratio to 10:11, specify:

        setsar=sar=10/11

*

To set a display aspect ratio of 16:9, and specify a maximum integer value of 1000 in the aspect ratio reduction, use the command:

        setdar=ratio=16/9:max=1000

setfield

Force field for the output video frame.

The "setfield" filter marks the interlace type field for the output frames. It does not change the input frame, but only sets the corresponding property, which affects how the frame is treated by following filters (e.g. "fieldorder" or "yadif").

The filter accepts the following options:

mode

Available values are:

auto: Keep the same field property.
bff: Mark the frame as bottom-field-first.
tff: Mark the frame as top-field-first.
prog: Mark the frame as progressive.

showinfo

Show a line containing various information for each input video frame. The input video is not modified.

The shown line contains a sequence of key/value pairs of the form key:value.

The following values are shown in the output:

n: The (sequential) number of the input frame, starting from 0.
pts: The Presentation TimeStamp of the input frame, expressed as a number of time base units. The time base unit depends on the filter input pad.
pts_time: The Presentation TimeStamp of the input frame, expressed as a number of seconds.
pos: The position of the frame in the input stream, or -1 if this information is unavailable and/or meaningless (for example in case of synthetic video).
fmt: The pixel format name.
sar: The sample aspect ratio of the input frame, expressed in the form num/den.
s: The size of the input frame. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual.
i: The type of interlaced mode (``P'' for ``progressive'', ``T'' for top field first, ``B'' for bottom field first).
iskey: This is 1 if the frame is a key frame, 0 otherwise.
type: The picture type of the input frame (``I'' for an I-frame, ``P'' for a P-frame, ``B'' for a B-frame, or ``?'' for an unknown type). Also refer to the documentation of the "AVPictureType" enum and of the "av_get_picture_type_char" function defined in libavutil/avutil.h.
checksum: The Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame.
plane_checksum: The Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, expressed in the form "[c0 c1 c2 c3]".

showpalette

Displays the 256 colors palette of each frame. This filter is only relevant for pal8 pixel format frames.

It accepts the following option:

s: Set the size of the box used to represent one palette color entry. Default is 30 (for a "30x30" pixel box).

shuffleframes

Reorder and/or duplicate video frames.

It accepts the following parameters:

mapping: Set the destination indexes of input frames. This is space or '|' separated list of indexes that maps input frames to output frames. Number of indexes also sets maximal value that each index may have.

The first frame has the index 0. The default is to keep the input unchanged.

Examples

*

Swap second and third frame of every three frames of the input:

        ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT

*

Swap 10th and 1st frame of every ten frames of the input:

        ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT

shuffleplanes

Reorder and/or duplicate video planes.

It accepts the following parameters:

map0: The index of the input plane to be used as the first output plane.
map1: The index of the input plane to be used as the second output plane.
map2: The index of the input plane to be used as the third output plane.
map3: The index of the input plane to be used as the fourth output plane.

The first plane has the index 0. The default is to keep the input unchanged.

Examples

*

Swap the second and third planes of the input:

        ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT

signalstats

Evaluate various visual metrics that assist in determining issues associated with the digitization of analog video media.

By default the filter will log these metadata values:

YMIN: Display the minimal Y value contained within the input frame. Expressed in range of [0-255].
YLOW: Display the Y value at the 10% percentile within the input frame. Expressed in range of [0-255].
YAVG: Display the average Y value within the input frame. Expressed in range of [0-255].
YHIGH: Display the Y value at the 90% percentile within the input frame. Expressed in range of [0-255].
YMAX: Display the maximum Y value contained within the input frame. Expressed in range of [0-255].
UMIN: Display the minimal U value contained within the input frame. Expressed in range of [0-255].
ULOW: Display the U value at the 10% percentile within the input frame. Expressed in range of [0-255].
UAVG: Display the average U value within the input frame. Expressed in range of [0-255].
UHIGH: Display the U value at the 90% percentile within the input frame. Expressed in range of [0-255].
UMAX: Display the maximum U value contained within the input frame. Expressed in range of [0-255].
VMIN: Display the minimal V value contained within the input frame. Expressed in range of [0-255].
VLOW: Display the V value at the 10% percentile within the input frame. Expressed in range of [0-255].
VAVG: Display the average V value within the input frame. Expressed in range of [0-255].
VHIGH: Display the V value at the 90% percentile within the input frame. Expressed in range of [0-255].
VMAX: Display the maximum V value contained within the input frame. Expressed in range of [0-255].
SATMIN: Display the minimal saturation value contained within the input frame. Expressed in range of [0-~181.02].
SATLOW: Display the saturation value at the 10% percentile within the input frame. Expressed in range of [0-~181.02].
SATAVG: Display the average saturation value within the input frame. Expressed in range of [0-~181.02].
SATHIGH: Display the saturation value at the 90% percentile within the input frame. Expressed in range of [0-~181.02].
SATMAX: Display the maximum saturation value contained within the input frame. Expressed in range of [0-~181.02].
HUEMED: Display the median value for hue within the input frame. Expressed in range of [0-360].
HUEAVG: Display the average value for hue within the input frame. Expressed in range of [0-360].
YDIF: Display the average of sample value difference between all values of the Y plane in the current frame and corresponding values of the previous input frame. Expressed in range of [0-255].
UDIF: Display the average of sample value difference between all values of the U plane in the current frame and corresponding values of the previous input frame. Expressed in range of [0-255].
VDIF: Display the average of sample value difference between all values of the V plane in the current frame and corresponding values of the previous input frame. Expressed in range of [0-255].
YBITDEPTH: Display bit depth of Y plane in current frame. Expressed in range of [0-16].
UBITDEPTH: Display bit depth of U plane in current frame. Expressed in range of [0-16].
VBITDEPTH: Display bit depth of V plane in current frame. Expressed in range of [0-16].

The filter accepts the following options:

stat

out

stat specify an additional form of image analysis. out output video with the specified type of pixel highlighted.

Both options accept the following values:

tout: Identify temporal outliers pixels. A temporal outlier is a pixel unlike the neighboring pixels of the same field. Examples of temporal outliers include the results of video dropouts, head clogs, or tape tracking issues.
vrep: Identify vertical line repetition. Vertical line repetition includes similar rows of pixels within a frame. In born-digital video vertical line repetition is common, but this pattern is uncommon in video digitized from an analog source. When it occurs in video that results from the digitization of an analog source it can indicate concealment from a dropout compensator.
brng: Identify pixels that fall outside of legal broadcast range.

color, c

Set the highlight color for the out option. The default color is yellow.

Examples

*

Output data of various video metrics:

        ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames

*

Output specific data about the minimum and maximum values of the Y plane per frame:

        ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN

*

Playback video while highlighting pixels that are outside of broadcast range in red.

        ffplay example.mov -vf signalstats="out=brng:color=red"

*

Playback video with signalstats metadata drawn over the frame.

        ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt

The contents of signalstat_drawtext.txt used in the command are:

        time %{pts:hms}
        Y (%{metadata:lavfi.signalstats.YMIN}-%{metadata:lavfi.signalstats.YMAX})
        U (%{metadata:lavfi.signalstats.UMIN}-%{metadata:lavfi.signalstats.UMAX})
        V (%{metadata:lavfi.signalstats.VMIN}-%{metadata:lavfi.signalstats.VMAX})
        saturation maximum: %{metadata:lavfi.signalstats.SATMAX}

smartblur

Blur the input video without impacting the outlines.

It accepts the following options:

luma_radius, lr: Set the luma radius. The option value must be a float number in the range [0.1,5.0] that specifies the variance of the gaussian filter used to blur the image (slower if larger). Default value is 1.0.
luma_strength, ls: Set the luma strength. The option value must be a float number in the range [-1.0,1.0] that configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included in [-1.0,0.0] will sharpen the image. Default value is 1.0.
luma_threshold, lt: Set the luma threshold used as a coefficient to determine whether a pixel should be blurred or not. The option value must be an integer in the range [-30,30]. A value of 0 will filter all the image, a value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges. Default value is 0.
chroma_radius, cr: Set the chroma radius. The option value must be a float number in the range [0.1,5.0] that specifies the variance of the gaussian filter used to blur the image (slower if larger). Default value is 1.0.
chroma_strength, cs: Set the chroma strength. The option value must be a float number in the range [-1.0,1.0] that configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included in [-1.0,0.0] will sharpen the image. Default value is 1.0.
chroma_threshold, ct: Set the chroma threshold used as a coefficient to determine whether a pixel should be blurred or not. The option value must be an integer in the range [-30,30]. A value of 0 will filter all the image, a value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges. Default value is 0.

If a chroma option is not explicitly set, the corresponding luma value is set.

ssim

Obtain the SSIM (Structural SImilarity Metric) between two input videos.

Both video inputs must have the same resolution and pixel format for this filter to work correctly. Also it assumes that both inputs have the same number of frames, which are compared one by one.

The filter stores the calculated SSIM of each frame.

The description of the accepted parameters follows.

stats_file, f: If specified the filter will use the named file to save the SSIM of each individual frame. When filename equals ``-'' the data is sent to standard output.

The file printed if stats_file is selected, contains a sequence of key/value pairs of the form key:value for each compared couple of frames.

A description of each shown parameter follows:

n: sequential number of the input frame, starting from 1
Y, U, V, R, G, B: SSIM of the compared frames for the component specified by the suffix.
All: SSIM of the compared frames for the whole frame.
dB: Same as above but in dB representation.

For example:

        movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
        [main][ref] ssim="stats_file=stats.log" [out]

On this example the input file being processed is compared with the reference file ref_movie.mpg. The SSIM of each individual frame is stored in stats.log.

Another example with both psnr and ssim at same time:

        ffmpeg -i main.mpg -i ref.mpg -lavfi  "ssim;[0:v][1:v]psnr" -f null -

stereo3d

Convert between different stereoscopic image formats.

The filters accept the following options:

in

Set stereoscopic image format of input.

Available values for input image formats are:

sbsl: side by side parallel (left eye left, right eye right)
sbsr: side by side crosseye (right eye left, left eye right)
sbs2l: side by side parallel with half width resolution (left eye left, right eye right)
sbs2r: side by side crosseye with half width resolution (right eye left, left eye right)
abl: above-below (left eye above, right eye below)
abr: above-below (right eye above, left eye below)
ab2l: above-below with half height resolution (left eye above, right eye below)
ab2r: above-below with half height resolution (right eye above, left eye below)
al: alternating frames (left eye first, right eye second)
ar: alternating frames (right eye first, left eye second)
irl: interleaved rows (left eye has top row, right eye starts on next row)
irr: interleaved rows (right eye has top row, left eye starts on next row)
icl: interleaved columns, left eye first
icr: interleaved columns, right eye first
Default value is sbsl.

out

Set stereoscopic image format of output.

sbsl: side by side parallel (left eye left, right eye right)
sbsr: side by side crosseye (right eye left, left eye right)
sbs2l: side by side parallel with half width resolution (left eye left, right eye right)
sbs2r: side by side crosseye with half width resolution (right eye left, left eye right)
abl: above-below (left eye above, right eye below)
abr: above-below (right eye above, left eye below)
ab2l: above-below with half height resolution (left eye above, right eye below)
ab2r: above-below with half height resolution (right eye above, left eye below)
al: alternating frames (left eye first, right eye second)
ar: alternating frames (right eye first, left eye second)
irl: interleaved rows (left eye has top row, right eye starts on next row)
irr: interleaved rows (right eye has top row, left eye starts on next row)
arbg: anaglyph red/blue gray (red filter on left eye, blue filter on right eye)
argg: anaglyph red/green gray (red filter on left eye, green filter on right eye)
arcg: anaglyph red/cyan gray (red filter on left eye, cyan filter on right eye)
arch: anaglyph red/cyan half colored (red filter on left eye, cyan filter on right eye)
arcc: anaglyph red/cyan color (red filter on left eye, cyan filter on right eye)
arcd: anaglyph red/cyan color optimized with the least squares projection of dubois (red filter on left eye, cyan filter on right eye)
agmg: anaglyph green/magenta gray (green filter on left eye, magenta filter on right eye)
agmh: anaglyph green/magenta half colored (green filter on left eye, magenta filter on right eye)
agmc: anaglyph green/magenta colored (green filter on left eye, magenta filter on right eye)
agmd: anaglyph green/magenta color optimized with the least squares projection of dubois (green filter on left eye, magenta filter on right eye)
aybg: anaglyph yellow/blue gray (yellow filter on left eye, blue filter on right eye)
aybh: anaglyph yellow/blue half colored (yellow filter on left eye, blue filter on right eye)
aybc: anaglyph yellow/blue colored (yellow filter on left eye, blue filter on right eye)
aybd: anaglyph yellow/blue color optimized with the least squares projection of dubois (yellow filter on left eye, blue filter on right eye)
ml: mono output (left eye only)
mr: mono output (right eye only)
chl: checkerboard, left eye first
chr: checkerboard, right eye first
icl: interleaved columns, left eye first
icr: interleaved columns, right eye first
hdmi: HDMI frame pack

: Default value is arcd.

Examples

*

Convert input video from side by side parallel to anaglyph yellow/blue dubois:

        stereo3d=sbsl:aybd

*

Convert input video from above below (left eye above, right eye below) to side by side crosseye.

        stereo3d=abl:sbsr

streamselect, astreamselect

Select video or audio streams.

The filter accepts the following options:

inputs: Set number of inputs. Default is 2.
map: Set input indexes to remap to outputs.

Commands

The "streamselect" and "astreamselect" filter supports the following commands:

map: Set input indexes to remap to outputs.

Examples

*

Select first 5 seconds 1st stream and rest of time 2nd stream:

        sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0

*

Same as above, but for audio:

        asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0

sobel

Apply sobel operator to input video stream.

The filter accepts the following option:

planes: Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
scale: Set value which will be multiplied with filtered result.
delta: Set value which will be added to filtered result.

spp

Apply a simple postprocessing filter that compresses and decompresses the image at several (or - in the case of quality level 6 - all) shifts and average the results.

The filter accepts the following options:

quality

Set quality. This option defines the number of levels for averaging. It accepts an integer in the range 0-6. If set to 0, the filter will have no effect. A value of 6 means the higher quality. For each increment of that value the speed drops by a factor of approximately 2. Default value is 3.

qp

Force a constant quantization parameter. If not set, the filter will use the QP from the video stream (if available).

mode

Set thresholding mode. Available modes are:

hard: Set hard thresholding (default).
soft: Set soft thresholding (better de-ringing effect, but likely blurrier).

use_bframe_qp

Enable the use of the QP from the B-Frames if set to 1. Using this option may cause flicker since the B-Frames have often larger QP. Default is 0 (not enabled).

subtitles

Draw subtitles on top of input video using the libass library.

To enable compilation of this filter you need to configure FFmpeg with "--enable-libass". This filter also requires a build with libavcodec and libavformat to convert the passed subtitles file to ASS (Advanced Substation Alpha) subtitles format.

The filter accepts the following options:

filename, f: Set the filename of the subtitle file to read. It must be specified.
original_size: Specify the size of the original video, the video for which the ASS file was composed. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Due to a misdesign in ASS aspect ratio arithmetic, this is necessary to correctly scale the fonts if the aspect ratio has been changed.
fontsdir: Set a directory path containing fonts that can be used by the filter. These fonts will be used in addition to whatever the font provider uses.
charenc: Set subtitles input character encoding. "subtitles" filter only. Only useful if not UTF-8.
stream_index, si: Set subtitles stream index. "subtitles" filter only.
force_style: Override default style or script info parameters of the subtitles. It accepts a string containing ASS style format "KEY=VALUE" couples separated by ``,''.

If the first key is not specified, it is assumed that the first value specifies the filename.

For example, to render the file sub.srt on top of the input video, use the command:

        subtitles=sub.srt

which is equivalent to:

        subtitles=filename=sub.srt

To render the default subtitles stream from file video.mkv, use:

        subtitles=video.mkv

To render the second subtitles stream from that file, use:

        subtitles=video.mkv:si=1

To make the subtitles stream from sub.srt appear in transparent green "DejaVu Serif", use:

        subtitles=sub.srt:force_style='FontName=DejaVu Serif,PrimaryColour=&HAA00FF00'

super2xsai

Scale the input by 2x and smooth using the Super2xSaI (Scale and Interpolate) pixel art scaling algorithm.

Useful for enlarging pixel art images without reducing sharpness.

swaprect

Swap two rectangular objects in video.

This filter accepts the following options:

w: Set object width.
h: Set object height.
x1: Set 1st rect x coordinate.
y1: Set 1st rect y coordinate.
x2: Set 2nd rect x coordinate.
y2: Set 2nd rect y coordinate.
All expressions are evaluated once for each frame.

The all options are expressions containing the following constants:

w
h: The input width and height.
a: same as w / h
sar: input sample aspect ratio
dar: input display aspect ratio, it is the same as (w / h) * sar
n: The number of the input frame, starting from 0.
t: The timestamp expressed in seconds. It's NAN if the input timestamp is unknown.
pos: the position in the file of the input frame, NAN if unknown

swapuv

Swap U & V plane.

telecine

Apply telecine process to the video.

This filter accepts the following options:

first_field

top, t: top field first
bottom, b: bottom field first The default value is "top".

pattern

A string of numbers representing the pulldown pattern you wish to apply. The default value is 23.

        Some typical patterns:
        
        NTSC output (30i):
        27.5p: 32222
        24p: 23 (classic)
        24p: 2332 (preferred)
        20p: 33
        18p: 334
        16p: 3444
        
        PAL output (25i):
        27.5p: 12222
        24p: 222222222223 ("Euro pulldown")
        16.67p: 33
        16p: 33333334

thumbnail

Select the most representative frame in a given sequence of consecutive frames.

The filter accepts the following options:

n: Set the frames batch size to analyze; in a set of n frames, the filter will pick one of them, and then handle the next batch of n frames until the end. Default is 100.

Since the filter keeps track of the whole frames sequence, a bigger n value will result in a higher memory usage, so a high value is not recommended.

Examples

*

Extract one picture each 50 frames:

        thumbnail=50

*

Complete example of a thumbnail creation with ffmpeg:

        ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png

tile

Tile several successive frames together.

The filter accepts the following options:

layout: Set the grid size (i.e. the number of lines and columns). For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual.
nb_frames: Set the maximum number of frames to render in the given area. It must be less than or equal to wxh. The default value is 0, meaning all the area will be used.
margin: Set the outer border margin in pixels.
padding: Set the inner border thickness (i.e. the number of pixels between frames). For more advanced padding options (such as having different values for the edges), refer to the pad video filter.
color: Specify the color of the unused area. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual. The default value of color is ``black''.

Examples

*

Produce 8x8 PNG tiles of all keyframes (-skip_frame nokey) in a movie:

        ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png

The -vsync 0 is necessary to prevent ffmpeg from duplicating each output frame to accommodate the originally detected frame rate.

*

Display 5 pictures in an area of "3x2" frames, with 7 pixels between them, and 2 pixels of initial margin, using mixed flat and named options:

        tile=3x2:nb_frames=5:padding=7:margin=2

tinterlace

Perform various types of temporal field interlacing.

Frames are counted starting from 1, so the first input frame is considered odd.

The filter accepts the following options:

mode

Specify the mode of the interlacing. This option can also be specified as a value alone. See below for a list of values for this option.

Available values are:

merge, 0

Move odd frames into the upper field, even into the lower field, generating a double height frame at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
        11111                           33333
        22222                           44444
        11111                           33333
        22222                           44444
        11111                           33333
        22222                           44444
        11111                           33333
        22222                           44444

drop_even, 1

Only output odd frames, even frames are dropped, generating a frame with unchanged height at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
        11111                           33333
        11111                           33333
        11111                           33333
        11111                           33333

drop_odd, 2

Only output even frames, odd frames are dropped, generating a frame with unchanged height at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
                        22222                           44444
                        22222                           44444
                        22222                           44444
                        22222                           44444

pad, 3

Expand each frame to full height, but pad alternate lines with black, generating a frame with double height at the same input frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
        11111           .....           33333           .....
        .....           22222           .....           44444
        11111           .....           33333           .....
        .....           22222           .....           44444
        11111           .....           33333           .....
        .....           22222           .....           44444
        11111           .....           33333           .....
        .....           22222           .....           44444

interleave_top, 4

Interleave the upper field from odd frames with the lower field from even frames, generating a frame with unchanged height at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111<-         22222           33333<-         44444
        11111           22222<-         33333           44444<-
        11111<-         22222           33333<-         44444
        11111           22222<-         33333           44444<-
        
        Output:
        11111                           33333
        22222                           44444
        11111                           33333
        22222                           44444

interleave_bottom, 5

Interleave the lower field from odd frames with the upper field from even frames, generating a frame with unchanged height at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222<-         33333           44444<-
        11111<-         22222           33333<-         44444
        11111           22222<-         33333           44444<-
        11111<-         22222           33333<-         44444
        
        Output:
        22222                           44444
        11111                           33333
        22222                           44444
        11111                           33333

interlacex2, 6

Double frame rate with unchanged height. Frames are inserted each containing the second temporal field from the previous input frame and the first temporal field from the next input frame. This mode relies on the top_field_first flag. Useful for interlaced video displays with no field synchronisation.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
         11111           22222           33333           44444
        11111           22222           33333           44444
         11111           22222           33333           44444
        
        Output:
        11111   22222   22222   33333   33333   44444   44444
         11111   11111   22222   22222   33333   33333   44444
        11111   22222   22222   33333   33333   44444   44444
         11111   11111   22222   22222   33333   33333   44444

mergex2, 7

Move odd frames into the upper field, even into the lower field, generating a double height frame at same frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
        11111           33333           33333           55555
        22222           22222           44444           44444
        11111           33333           33333           55555
        22222           22222           44444           44444
        11111           33333           33333           55555
        22222           22222           44444           44444
        11111           33333           33333           55555
        22222           22222           44444           44444

Numeric values are deprecated but are accepted for backward compatibility reasons.

Default mode is "merge".

flags

Specify flags influencing the filter process.

Available value for flags is:

low_pass_filter, vlfp: Enable vertical low-pass filtering in the filter. Vertical low-pass filtering is required when creating an interlaced destination from a progressive source which contains high-frequency vertical detail. Filtering will reduce interlace 'twitter' and Moire patterning.
Vertical low-pass filtering can only be enabled for mode interleave_top and interleave_bottom.

transpose

Transpose rows with columns in the input video and optionally flip it.

It accepts the following parameters:

dir

Specify the transposition direction.

Can assume the following values:

0, 4, cclock_flip

Rotate by 90 degrees counterclockwise and vertically flip (default), that is:

        L.R     L.l
        . . ->  . .
        l.r     R.r

1, 5, clock

Rotate by 90 degrees clockwise, that is:

        L.R     l.L
        . . ->  . .
        l.r     r.R

2, 6, cclock

Rotate by 90 degrees counterclockwise, that is:

        L.R     R.r
        . . ->  . .
        l.r     L.l

3, 7, clock_flip

Rotate by 90 degrees clockwise and vertically flip, that is:

        L.R     r.R
        . . ->  . .
        l.r     l.L

For values between 4-7, the transposition is only done if the input video geometry is portrait and not landscape. These values are deprecated, the "passthrough" option should be used instead.

Numerical values are deprecated, and should be dropped in favor of symbolic constants.

passthrough

Do not apply the transposition if the input geometry matches the one specified by the specified value. It accepts the following values:

none: Always apply transposition.
portrait: Preserve portrait geometry (when height >= width).
landscape: Preserve landscape geometry (when width >= height).

: Default value is "none".

For example to rotate by 90 degrees clockwise and preserve portrait layout:

        transpose=dir=1:passthrough=portrait

The command above can also be specified as:

        transpose=1:portrait

trim

Trim the input so that the output contains one continuous subpart of the input.

It accepts the following parameters:

start: Specify the time of the start of the kept section, i.e. the frame with the timestamp start will be the first frame in the output.
end: Specify the time of the first frame that will be dropped, i.e. the frame immediately preceding the one with the timestamp end will be the last frame in the output.
start_pts: This is the same as start, except this option sets the start timestamp in timebase units instead of seconds.
end_pts: This is the same as end, except this option sets the end timestamp in timebase units instead of seconds.
duration: The maximum duration of the output in seconds.
start_frame: The number of the first frame that should be passed to the output.
end_frame: The number of the first frame that should be dropped.

start, end, and duration are expressed as time duration specifications; see the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax.

Note that the first two sets of the start/end options and the duration option look at the frame timestamp, while the _frame variants simply count the frames that pass through the filter. Also note that this filter does not modify the timestamps. If you wish for the output timestamps to start at zero, insert a setpts filter after the trim filter.

If multiple start or end options are set, this filter tries to be greedy and keep all the frames that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple trim filters.

The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time.

Examples:

*

Drop everything except the second minute of input:

        ffmpeg -i INPUT -vf trim=60:120

*

Keep only the first second:

        ffmpeg -i INPUT -vf trim=duration=1

unsharp

Sharpen or blur the input video.

It accepts the following parameters:

luma_msize_x, lx

Set the luma matrix horizontal size. It must be an odd integer between 3 and 23. The default value is 5.

luma_msize_y, ly

Set the luma matrix vertical size. It must be an odd integer between 3 and 23. The default value is 5.

luma_amount, la

Set the luma effect strength. It must be a floating point number, reasonable values lay between -1.5 and 1.5.

Negative values will blur the input video, while positive values will sharpen it, a value of zero will disable the effect.

Default value is 1.0.

chroma_msize_x, cx

Set the chroma matrix horizontal size. It must be an odd integer between 3 and 23. The default value is 5.

chroma_msize_y, cy

Set the chroma matrix vertical size. It must be an odd integer between 3 and 23. The default value is 5.

chroma_amount, ca

Set the chroma effect strength. It must be a floating point number, reasonable values lay between -1.5 and 1.5.

Negative values will blur the input video, while positive values will sharpen it, a value of zero will disable the effect.

Default value is 0.0.

opencl

If set to 1, specify using OpenCL capabilities, only available if FFmpeg was configured with "--enable-opencl". Default value is 0.

All parameters are optional and default to the equivalent of the string '5:5:1.0:5:5:0.0'.

Examples

*

Apply strong luma sharpen effect:

        unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5

*

Apply a strong blur of both luma and chroma parameters:

        unsharp=7:7:-2:7:7:-2

uspp

Apply ultra slow/simple postprocessing filter that compresses and decompresses the image at several (or - in the case of quality level 8 - all) shifts and average the results.

The way this differs from the behavior of spp is that uspp actually encodes & decodes each case with libavcodec Snow, whereas spp uses a simplified intra only 8x8 DCT similar to MJPEG.

The filter accepts the following options:

quality: Set quality. This option defines the number of levels for averaging. It accepts an integer in the range 0-8. If set to 0, the filter will have no effect. A value of 8 means the higher quality. For each increment of that value the speed drops by a factor of approximately 2. Default value is 3.
qp: Force a constant quantization parameter. If not set, the filter will use the QP from the video stream (if available).

vaguedenoiser

Apply a wavelet based denoiser.

It transforms each frame from the video input into the wavelet domain, using Cohen-Daubechies-Feauveau 9/7. Then it applies some filtering to the obtained coefficients. It does an inverse wavelet transform after. Due to wavelet properties, it should give a nice smoothed result, and reduced noise, without blurring picture features.

This filter accepts the following options:

threshold

The filtering strength. The higher, the more filtered the video will be. Hard thresholding can use a higher threshold than soft thresholding before the video looks overfiltered.

method

The filtering method the filter will use.

It accepts the following values:

hard: All values under the threshold will be zeroed.
soft: All values under the threshold will be zeroed. All values above will be reduced by the threshold.
garrote: Scales or nullifies coefficients - intermediary between (more) soft and (less) hard thresholding.

nsteps

Number of times, the wavelet will decompose the picture. Picture can't be decomposed beyond a particular point (typically, 8 for a 640x480 frame - as 2^9 = 512 > 480)

percent

Partial of full denoising (limited coefficients shrinking), from 0 to 100.

planes

A list of the planes to process. By default all planes are processed.

vectorscope

Display 2 color component values in the two dimensional graph (which is called a vectorscope).

This filter accepts the following options:

mode, m

Set vectorscope mode.

It accepts the following values:

gray: Gray values are displayed on graph, higher brightness means more pixels have same component color value on location in graph. This is the default mode.
color: Gray values are displayed on graph. Surrounding pixels values which are not present in video frame are drawn in gradient of 2 color components which are set by option "x" and "y". The 3rd color component is static.
color2: Actual color components values present in video frame are displayed on graph.
color3: Similar as color2 but higher frequency of same values "x" and "y" on graph increases value of another color component, which is luminance by default values of "x" and "y".
color4: Actual colors present in video frame are displayed on graph. If two different colors map to same position on graph then color with higher value of component not present in graph is picked.
color5: Gray values are displayed on graph. Similar to "color" but with 3rd color component picked from radial gradient.

x

Set which color component will be represented on X-axis. Default is 1.

y

Set which color component will be represented on Y-axis. Default is 2.

intensity, i

Set intensity, used by modes: gray, color, color3 and color5 for increasing brightness of color component which represents frequency of (X, Y) location in graph.

envelope, e

none: No envelope, this is default.
instant: Instant envelope, even darkest single pixel will be clearly highlighted.
peak: Hold maximum and minimum values presented in graph over time. This way you can still spot out of range values without constantly looking at vectorscope.
peak+instant: Peak and instant envelope combined together.

graticule, g

Set what kind of graticule to draw.

none
green
color

opacity, o

Set graticule opacity.

flags, f

Set graticule flags.

white: Draw graticule for white point.
black: Draw graticule for black point.
name: Draw color points short names.

bgopacity, b

Set background opacity.

lthreshold, l

Set low threshold for color component not represented on X or Y axis. Values lower than this value will be ignored. Default is 0. Note this value is multiplied with actual max possible value one pixel component can have. So for 8-bit input and low threshold value of 0.1 actual threshold is 0.1 * 255 = 25.

hthreshold, h

Set high threshold for color component not represented on X or Y axis. Values higher than this value will be ignored. Default is 1. Note this value is multiplied with actual max possible value one pixel component can have. So for 8-bit input and high threshold value of 0.9 actual threshold is 0.9 * 255 = 230.

colorspace, c

Set what kind of colorspace to use when drawing graticule.

auto
601
709

: Default is auto.

vidstabdetect

Analyze video stabilization/deshaking. Perform pass 1 of 2, see vidstabtransform for pass 2.

This filter generates a file with relative translation and rotation transform information about subsequent frames, which is then used by the vidstabtransform filter.

To enable compilation of this filter you need to configure FFmpeg with "--enable-libvidstab".

This filter accepts the following options:

result

Set the path to the file used to write the transforms information. Default value is transforms.trf.

shakiness

Set how shaky the video is and how quick the camera is. It accepts an integer in the range 1-10, a value of 1 means little shakiness, a value of 10 means strong shakiness. Default value is 5.

accuracy

Set the accuracy of the detection process. It must be a value in the range 1-15. A value of 1 means low accuracy, a value of 15 means high accuracy. Default value is 15.

stepsize

Set stepsize of the search process. The region around minimum is scanned with 1 pixel resolution. Default value is 6.

mincontrast

Set minimum contrast. Below this value a local measurement field is discarded. Must be a floating point value in the range 0-1. Default value is 0.3.

tripod

Set reference frame number for tripod mode.

If enabled, the motion of the frames is compared to a reference frame in the filtered stream, identified by the specified number. The idea is to compensate all movements in a more-or-less static scene and keep the camera view absolutely still.

If set to 0, it is disabled. The frames are counted starting from 1.

show

Show fields and transforms in the resulting frames. It accepts an integer in the range 0-2. Default value is 0, which disables any visualization.

Examples

*

Use default values:

        vidstabdetect

*

Analyze strongly shaky movie and put the results in file mytransforms.trf:

        vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"

*

Visualize the result of internal transformations in the resulting video:

        vidstabdetect=show=1

*

Analyze a video with medium shakiness using ffmpeg:

        ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi

vidstabtransform

Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass 1.

Read a file with transform information for each frame and apply/compensate them. Together with the vidstabdetect filter this can be used to deshake videos. See also <http://public.hronopik.de/vid.stab>. It is important to also use the unsharp filter, see below.

To enable compilation of this filter you need to configure FFmpeg with "--enable-libvidstab".

Options

input

Set path to the file used to read the transforms. Default value is transforms.trf.

smoothing

Set the number of frames (value*2 + 1) used for lowpass filtering the camera movements. Default value is 10.

For example a number of 10 means that 21 frames are used (10 in the past and 10 in the future) to smoothen the motion in the video. A larger value leads to a smoother video, but limits the acceleration of the camera (pan/tilt movements). 0 is a special case where a static camera is simulated.

optalgo

Set the camera path optimization algorithm.

Accepted values are:

gauss: gaussian kernel low-pass filter on camera motion (default)
avg: averaging on transformations

maxshift

Set maximal number of pixels to translate frames. Default value is -1, meaning no limit.

maxangle

Set maximal angle in radians (degree*PI/180) to rotate frames. Default value is -1, meaning no limit.

crop

Specify how to deal with borders that may be visible due to movement compensation.

Available values are:

keep: keep image information from previous frame (default)
black: fill the border black

invert

Invert transforms if set to 1. Default value is 0.

relative

Consider transforms as relative to previous frame if set to 1, absolute if set to 0. Default value is 0.

zoom

Set percentage to zoom. A positive value will result in a zoom-in effect, a negative value in a zoom-out effect. Default value is 0 (no zoom).

optzoom

Set optimal zooming to avoid borders.

Accepted values are:

0: disabled
1: optimal static zoom value is determined (only very strong movements will lead to visible borders) (default)
2: optimal adaptive zoom value is determined (no borders will be visible), see zoomspeed

: Note that the value given at zoom is added to the one calculated here.

zoomspeed

Set percent to zoom maximally each frame (enabled when optzoom is set to 2). Range is from 0 to 5, default value is 0.25.

interpol

Specify type of interpolation.

Available values are:

no: no interpolation
linear: linear only horizontal
bilinear: linear in both directions (default)
bicubic: cubic in both directions (slow)

tripod

Enable virtual tripod mode if set to 1, which is equivalent to "relative=0:smoothing=0". Default value is 0.

Use also "tripod" option of vidstabdetect.

debug

Increase log verbosity if set to 1. Also the detected global motions are written to the temporary file global_motions.trf. Default value is 0.

Examples

*

Use ffmpeg for a typical stabilization with default values:

        ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg

Note the use of the unsharp filter which is always recommended.

*

Zoom in a bit more and load transform data from a given file:

        vidstabtransform=zoom=5:input="mytransforms.trf"

*

Smoothen the video even more:

        vidstabtransform=smoothing=30

vflip

Flip the input video vertically.

For example, to vertically flip a video with ffmpeg:

        ffmpeg -i in.avi -vf "vflip" out.avi

vignette

Make or reverse a natural vignetting effect.

The filter accepts the following options:

angle, a

Set lens angle expression as a number of radians.

The value is clipped in the "[0,PI/2]" range.

Default value: "PI/5"

x0

y0

Set center coordinates expressions. Respectively "w/2" and "h/2" by default.

mode

Set forward/backward mode.

Available modes are:

forward: The larger the distance from the central point, the darker the image becomes.
backward: The larger the distance from the central point, the brighter the image becomes. This can be used to reverse a vignette effect, though there is no automatic detection to extract the lens angle and other settings (yet). It can also be used to create a burning effect.

: Default value is forward.

eval

Set evaluation mode for the expressions (angle, x0, y0).

It accepts the following values:

init: Evaluate expressions only once during the filter initialization.
frame: Evaluate expressions for each incoming frame. This is way slower than the init mode since it requires all the scalers to be re-computed, but it allows advanced dynamic expressions.

: Default value is init.

dither

Set dithering to reduce the circular banding effects. Default is 1 (enabled).

aspect

Set vignette aspect. This setting allows one to adjust the shape of the vignette. Setting this value to the SAR of the input will make a rectangular vignetting following the dimensions of the video.

Default is "1/1".

Expressions

The alpha, x0 and y0 expressions can contain the following parameters.

w
h: input width and height
n: the number of input frame, starting from 0
pts: the PTS (Presentation TimeStamp) time of the filtered video frame, expressed in TB units, NAN if undefined
r: frame rate of the input video, NAN if the input frame rate is unknown
t: the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined
tb: time base of the input video

Examples

*

Apply simple strong vignetting effect:

        vignette=PI/4

*

Make a flickering vignetting:

        vignette='PI/4+random(1)*PI/50':eval=frame

vstack

Stack input videos vertically.

All streams must be of same pixel format and of same width.

Note that this filter is faster than using overlay and pad filter to create same output.

The filter accept the following option:

inputs: Set number of input streams. Default is 2.
shortest: If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.

w3fdif

Deinterlace the input video (``w3fdif'' stands for ``Weston 3 Field Deinterlacing Filter'').

Based on the process described by Martin Weston for BBC R&D, and implemented based on the de-interlace algorithm written by Jim Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter uses filter coefficients calculated by BBC R&D.

There are two sets of filter coefficients, so called ``simple'': and ``complex''. Which set of filter coefficients is used can be set by passing an optional parameter:

filter

Set the interlacing filter coefficients. Accepts one of the following values:

simple: Simple filter coefficient set.
complex: More-complex filter coefficient set.

: Default value is complex.

deint

Specify which frames to deinterlace. Accept one of the following values:

all: Deinterlace all frames,
interlaced: Only deinterlace frames marked as interlaced.

: Default value is all.

waveform

Video waveform monitor.

The waveform monitor plots color component intensity. By default luminance only. Each column of the waveform corresponds to a column of pixels in the source video.

It accepts the following options:

mode, m

Can be either "row", or "column". Default is "column". In row mode, the graph on the left side represents color component value 0 and the right side represents value = 255. In column mode, the top side represents color component value = 0 and bottom side represents value = 255.

intensity, i

Set intensity. Smaller values are useful to find out how many values of the same luminance are distributed across input rows/columns. Default value is 0.04. Allowed range is [0, 1].

mirror, r

Set mirroring mode. 0 means unmirrored, 1 means mirrored. In mirrored mode, higher values will be represented on the left side for "row" mode and at the top for "column" mode. Default is 1 (mirrored).

display, d

Set display mode. It accepts the following values:

overlay: Presents information identical to that in the "parade", except that the graphs representing color components are superimposed directly over one another.
This display mode makes it easier to spot relative differences or similarities in overlapping areas of the color components that are supposed to be identical, such as neutral whites, grays, or blacks.
stack: Display separate graph for the color components side by side in "row" mode or one below the other in "column" mode.
parade: Display separate graph for the color components side by side in "column" mode or one below the other in "row" mode.
Using this display mode makes it easy to spot color casts in the highlights and shadows of an image, by comparing the contours of the top and the bottom graphs of each waveform. Since whites, grays, and blacks are characterized by exactly equal amounts of red, green, and blue, neutral areas of the picture should display three waveforms of roughly equal width/height. If not, the correction is easy to perform by making level adjustments the three waveforms.

: Default is "stack".

components, c

Set which color components to display. Default is 1, which means only luminance or red color component if input is in RGB colorspace. If is set for example to 7 it will display all 3 (if) available color components.

envelope, e

none: No envelope, this is default.
instant: Instant envelope, minimum and maximum values presented in graph will be easily visible even with small "step" value.
peak: Hold minimum and maximum values presented in graph across time. This way you can still spot out of range values without constantly looking at waveforms.
peak+instant: Peak and instant envelope combined together.

filter, f

lowpass: No filtering, this is default.
flat: Luma and chroma combined together.
aflat: Similar as above, but shows difference between blue and red chroma.
chroma: Displays only chroma.
color: Displays actual color value on waveform.
acolor: Similar as above, but with luma showing frequency of chroma values.

graticule, g

Set which graticule to display.

none: Do not display graticule.
green: Display green graticule showing legal broadcast ranges.

opacity, o

Set graticule opacity.

flags, fl

Set graticule flags.

numbers: Draw numbers above lines. By default enabled.
dots: Draw dots instead of lines.

scale, s

Set scale used for displaying graticule.

digital
millivolts
ire

: Default is digital.

bgopacity, b

Set background opacity.

weave

The "weave" takes a field-based video input and join each two sequential fields into single frame, producing a new double height clip with half the frame rate and half the frame count.

It accepts the following option:

first_field

Set first field. Available values are:

top, t: Set the frame as top-field-first.
bottom, b: Set the frame as bottom-field-first.

Examples

*

Interlace video using select and separatefields filter:

        separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave

xbr

Apply the xBR high-quality magnification filter which is designed for pixel art. It follows a set of edge-detection rules, see <http://www.libretro.com/forums/viewtopic.php?f=6&t=134>.

It accepts the following option:

n: Set the scaling dimension: 2 for "2xBR", 3 for "3xBR" and 4 for "4xBR". Default is 3.

yadif

Deinterlace the input video (``yadif'' means ``yet another deinterlacing filter'').

It accepts the following parameters:

mode

The interlacing mode to adopt. It accepts one of the following values:

0, send_frame: Output one frame for each frame.
1, send_field: Output one frame for each field.
2, send_frame_nospatial: Like "send_frame", but it skips the spatial interlacing check.
3, send_field_nospatial: Like "send_field", but it skips the spatial interlacing check.

: The default value is "send_frame".

parity

The picture field parity assumed for the input interlaced video. It accepts one of the following values:

0, tff: Assume the top field is first.
1, bff: Assume the bottom field is first.
-1, auto: Enable automatic detection of field parity.

: The default value is "auto". If the interlacing is unknown or the decoder does not export this information, top field first will be assumed.

deint

Specify which frames to deinterlace. Accept one of the following values:

0, all: Deinterlace all frames.
1, interlaced: Only deinterlace frames marked as interlaced.

: The default value is "all".

zoompan

Apply Zoom & Pan effect.

This filter accepts the following options:

zoom, z: Set the zoom expression. Default is 1.
x
y: Set the x and y expression. Default is 0.
d: Set the duration expression in number of frames. This sets for how many number of frames effect will last for single input image.
s: Set the output image size, default is 'hd720'.
fps: Set the output frame rate, default is '25'.

Each expression can contain the following constants:

in_w, iw: Input width.
in_h, ih: Input height.
out_w, ow: Output width.
out_h, oh: Output height.
in: Input frame count.
on: Output frame count.
x
y: Last calculated 'x' and 'y' position from 'x' and 'y' expression for current input frame.
px
py: 'x' and 'y' of last output frame of previous input frame or 0 when there was not yet such frame (first input frame).
zoom: Last calculated zoom from 'z' expression for current input frame.
pzoom: Last calculated zoom of last output frame of previous input frame.
duration: Number of output frames for current input frame. Calculated from 'd' expression for each input frame.
pduration: number of output frames created for previous input frame
a: Rational number: input width / input height
sar: sample aspect ratio
dar: display aspect ratio

Examples

*

Zoom-in up to 1.5 and pan at same time to some spot near center of picture:

        zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360

*

Zoom-in up to 1.5 and pan always at center of picture:

        zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

*

Same as above but without pausing:

        zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

zscale

Scale (resize) the input video, using the z.lib library: https://github.com/sekrit-twc/zimg.

The zscale filter forces the output display aspect ratio to be the same as the input, by changing the output sample aspect ratio.

If the input image format is different from the format requested by the next filter, the zscale filter will convert the input to the requested format.

Options

The filter accepts the following options.

width, w

height, h

Set the output video dimension expression. Default value is the input dimension.

If the width or w is 0, the input width is used for the output. If the height or h is 0, the input height is used for the output.

If one of the values is -1, the zscale filter will use a value that maintains the aspect ratio of the input image, calculated from the other specified dimension. If both of them are -1, the input size is used

If one of the values is -n with n > 1, the zscale filter will also use a value that maintains the aspect ratio of the input image, calculated from the other specified dimension. After that it will, however, make sure that the calculated dimension is divisible by n and adjust the value if necessary.

See below for the list of accepted constants for use in the dimension expression.

size, s

Set the video size. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual.

dither, d

Set the dither type.

Possible values are:

none
ordered
random
error_diffusion

: Default is none.

filter, f

Set the resize filter type.

Possible values are:

point
bilinear
bicubic
spline16
spline36
lanczos

: Default is bilinear.

range, r

Set the color range.

Possible values are:

input
limited
full

: Default is same as input.

primaries, p

Set the color primaries.

Possible values are:

input
709
unspecified
170m
240m
2020

: Default is same as input.

transfer, t

Set the transfer characteristics.

Possible values are:

input
709
unspecified
601
linear
2020_10
2020_12

: Default is same as input.

matrix, m

Set the colorspace matrix.

Possible value are:

input
709
unspecified
470bg
170m
2020_ncl
2020_cl

: Default is same as input.

rangein, rin

Set the input color range.

Possible values are:

input
limited
full

: Default is same as input.

primariesin, pin

Set the input color primaries.

Possible values are:

input
709
unspecified
170m
240m
2020

: Default is same as input.

transferin, tin

Set the input transfer characteristics.

Possible values are:

input
709
unspecified
601
linear
2020_10
2020_12

: Default is same as input.

matrixin, min

Set the input colorspace matrix.

Possible value are:

input
709
unspecified
470bg
170m
2020_ncl
2020_cl

chromal, c

Set the output chroma location.

Possible values are:

input
left
center
topleft
top
bottomleft
bottom

chromalin, cin

Set the input chroma location.

Possible values are:

input
left
center
topleft
top
bottomleft
bottom

The values of the w and h options are expressions containing the following constants:

in_w
in_h: The input width and height
iw
ih: These are the same as in_w and in_h.
out_w
out_h: The output (scaled) width and height
ow
oh: These are the same as out_w and out_h
a: The same as iw / ih
sar: input sample aspect ratio
dar: The input display aspect ratio. Calculated from "(iw / ih) * sar".
hsub
vsub: horizontal and vertical input chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.
ohsub
ovsub: horizontal and vertical output chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

VIDEO SOURCES

Below is a description of the currently available video sources.

buffer

Buffer video frames, and make them available to the filter chain.

This source is mainly intended for a programmatic use, in particular through the interface defined in libavfilter/vsrc_buffer.h.

It accepts the following parameters:

video_size: Specify the size (width and height) of the buffered video frames. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual.
width: The input video width.
height: The input video height.
pix_fmt: A string representing the pixel format of the buffered video frames. It may be a number corresponding to a pixel format, or a pixel format name.
time_base: Specify the timebase assumed by the timestamps of the buffered frames.
frame_rate: Specify the frame rate expected for the video stream.
pixel_aspect, sar: The sample (pixel) aspect ratio of the input video.
sws_param: Specify the optional parameters to be used for the scale filter which is automatically inserted when an input change is detected in the input size or format.
hw_frames_ctx: When using a hardware pixel format, this should be a reference to an AVHWFramesContext describing input frames.

For example:

        buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1

will instruct the source to accept video frames with size 320x240 and with format ``yuv410p'', assuming 1/24 as the timestamps timebase and square pixels (1:1 sample aspect ratio). Since the pixel format with name ``yuv410p'' corresponds to the number 6 (check the enum AVPixelFormat definition in libavutil/pixfmt.h), this example corresponds to:

        buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1

Alternatively, the options can be specified as a flat string, but this syntax is deprecated:

width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den[:sws_param]

cellauto

Create a pattern generated by an elementary cellular automaton.

The initial state of the cellular automaton can be defined through the filename and pattern options. If such options are not specified an initial state is created randomly.

At each new frame a new row in the video is filled with the result of the cellular automaton next generation. The behavior when the whole frame is filled is defined by the scroll option.

This source accepts the following options:

filename, f

Read the initial cellular automaton state, i.e. the starting row, from the specified file. In the file, each non-whitespace character is considered an alive cell, a newline will terminate the row, and further characters in the file will be ignored.

pattern, p

Read the initial cellular automaton state, i.e. the starting row, from the specified string.

Each non-whitespace character in the string is considered an alive cell, a newline will terminate the row, and further characters in the string will be ignored.

rate, r

Set the video rate, that is the number of frames generated per second. Default is 25.

random_fill_ratio, ratio

Set the random fill ratio for the initial cellular automaton row. It is a floating point number value ranging from 0 to 1, defaults to 1/PHI.

This option is ignored when a file or a pattern is specified.

random_seed, seed

Set the seed for filling randomly the initial row, must be an integer included between 0 and UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random seed on a best effort basis.

rule

Set the cellular automaton rule, it is a number ranging from 0 to 255. Default value is 110.

size, s

Set the size of the output video. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual.

If filename or pattern is specified, the size is set by default to the width of the specified initial state row, and the height is set to width * PHI.

If size is set, it must contain the width of the specified pattern string, and the specified pattern will be centered in the larger row.

If a filename or a pattern string is not specified, the size value defaults to ``320x518'' (used for a randomly generated initial state).

scroll

If set to 1, scroll the output upward when all the rows in the output have been already filled. If set to 0, the new generated row will be written over the top row just after the bottom row is filled. Defaults to 1.

start_full, full

If set to 1, completely fill the output with generated rows before outputting the first frame. This is the default behavior, for disabling set the value to 0.

stitch

If set to 1, stitch the left and right row edges together. This is the default behavior, for disabling set the value to 0.

Examples

*

Read the initial state from pattern, and specify an output of size 200x400.

        cellauto=f=pattern:s=200x400

*

Generate a random initial row with a width of 200 cells, with a fill ratio of 2/3:

        cellauto=ratio=2/3:s=200x200

*

Create a pattern generated by rule 18 starting by a single alive cell centered on an initial row with width 100:

        cellauto=p=@s=100x400:full=0:rule=18

*

Specify a more elaborated initial pattern:

        cellauto=p='@@ @ @@':s=100x400:full=0:rule=18

coreimagesrc

Video source generated on GPU using Apple's CoreImage API on OSX.

This video source is a specialized version of the coreimage video filter. Use a core image generator at the beginning of the applied filterchain to generate the content.

The coreimagesrc video source accepts the following options:

list_generators

List all available generators along with all their respective options as well as possible minimum and maximum values along with the default values.

        list_generators=true

size, s

Specify the size of the sourced video. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. The default value is "320x240".

rate, r

Specify the frame rate of the sourced video, as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a floating point number or a valid video frame rate abbreviation. The default value is ``25''.

sar

Set the sample aspect ratio of the sourced video.

duration, d

Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax.

If not specified, or the expressed duration is negative, the video is supposed to be generated forever.

Additionally, all options of the coreimage video filter are accepted. A complete filterchain can be used for further processing of the generated input without CPU-HOST transfer. See coreimage documentation and examples for details.

Examples

*

Use CIQRCodeGenerator to create a QR code for the FFmpeg homepage, given as complete and escaped command-line for Apple's standard bash shell:

        ffmpeg -f lavfi -i coreimagesrc=s=100x100:filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

This example is equivalent to the QRCode example of coreimage without the need for a nullsrc video source.

mandelbrot

Generate a Mandelbrot set fractal, and progressively zoom towards the point specified with start_x and start_y.

This source accepts the following options:

end_pts

Set the terminal pts value. Default value is 400.

end_scale

Set the terminal scale value. Must be a floating point value. Default value is 0.3.

inner

Set the inner coloring mode, that is the algorithm used to draw the Mandelbrot fractal internal region.

It shall assume one of the following values:

black: Set black mode.
convergence: Show time until convergence.
mincol: Set color based on point closest to the origin of the iterations.
period: Set period mode.

: Default value is mincol.

bailout

Set the bailout value. Default value is 10.0.

maxiter

Set the maximum of iterations performed by the rendering algorithm. Default value is 7189.

outer

Set outer coloring mode. It shall assume one of following values:

iteration_count: Set iteration cound mode.
normalized_iteration_count: set normalized iteration count mode.

: Default value is normalized_iteration_count.

rate, r

Set frame rate, expressed as number of frames per second. Default value is ``25''.

size, s

Set frame size. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default value is ``640x480''.

start_scale

Set the initial scale value. Default value is 3.0.

start_x

Set the initial x position. Must be a floating point value between -100 and 100. Default value is -0.743643887037158704752191506114774.

start_y

Set the initial y position. Must be a floating point value between -100 and 100. Default value is -0.131825904205311970493132056385139.

mptestsrc

Generate various test patterns, as generated by the MPlayer test filter.

The size of the generated video is fixed, and is 256x256. This source is useful in particular for testing encoding features.

This source accepts the following options:

rate, r

duration, d

Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax.

If not specified, or the expressed duration is negative, the video is supposed to be generated forever.

test, t

Set the number or the name of the test to perform. Supported tests are:

dc_luma
dc_chroma
freq_luma
freq_chroma
amp_luma
amp_chroma
cbp
mv
ring1
ring2
all

: Default value is ``all'', which will cycle through the list of all tests.

Some examples:

        mptestsrc=t=dc_luma

will generate a ``dc_luma'' test pattern.

frei0r_src

Provide a frei0r source.

To enable compilation of this filter you need to install the frei0r header and configure FFmpeg with "--enable-frei0r".

This source accepts the following parameters:

size: The size of the video to generate. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual.
framerate: The framerate of the generated video. It may be a string of the form num/den or a frame rate abbreviation.
filter_name: The name to the frei0r source to load. For more information regarding frei0r and how to set the parameters, read the frei0r section in the video filters documentation.
filter_params: A '|'-separated list of parameters to pass to the frei0r source.

For example, to generate a frei0r partik0l source with size 200x200 and frame rate 10 which is overlaid on the overlay filter main input:

        frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay

life

Generate a life pattern.

This source is based on a generalization of John Conway's life game.

The sourced input represents a life grid, each pixel represents a cell which can be in one of two possible states, alive or dead. Every cell interacts with its eight neighbours, which are the cells that are horizontally, vertically, or diagonally adjacent.

At each interaction the grid evolves according to the adopted rule, which specifies the number of neighbor alive cells which will make a cell stay alive or born. The rule option allows one to specify the rule to adopt.

This source accepts the following options:

filename, f

Set the file from which to read the initial grid state. In the file, each non-whitespace character is considered an alive cell, and newline is used to delimit the end of each row.

If this option is not specified, the initial grid is generated randomly.

rate, r

Set the video rate, that is the number of frames generated per second. Default is 25.

random_fill_ratio, ratio

Set the random fill ratio for the initial random grid. It is a floating point number value ranging from 0 to 1, defaults to 1/PHI. It is ignored when a file is specified.

random_seed, seed

Set the seed for filling the initial random grid, must be an integer included between 0 and UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random seed on a best effort basis.

rule

Set the life rule.

A rule can be specified with a code of the kind "SNS/BNB", where NS and NB are sequences of numbers in the range 0-8, NS specifies the number of alive neighbor cells which make a live cell stay alive, and NB the number of alive neighbor cells which make a dead cell to become alive (i.e. to ``born''). ``s'' and ``b'' can be used in place of ``S'' and ``B'', respectively.

Alternatively a rule can be specified by an 18-bits integer. The 9 high order bits are used to encode the next cell state if it is alive for each number of neighbor alive cells, the low order bits specify the rule for ``borning'' new cells. Higher order bits encode for an higher number of neighbor cells. For example the number 6153 = "(12<<9)+9" specifies a stay alive rule of 12 and a born rule of 9, which corresponds to ``S23/B03''.

Default value is ``S23/B3'', which is the original Conway's game of life rule, and will keep a cell alive if it has 2 or 3 neighbor alive cells, and will born a new cell if there are three alive cells around a dead cell.

size, s

Set the size of the output video. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual.

If filename is specified, the size is set by default to the same size of the input file. If size is set, it must contain the size specified in the input file, and the initial grid defined in that file is centered in the larger resulting area.

If a filename is not specified, the size value defaults to ``320x240'' (used for a randomly generated initial grid).

stitch

If set to 1, stitch the left and right grid edges together, and the top and bottom edges also. Defaults to 1.

mold

Set cell mold speed. If set, a dead cell will go from death_color to mold_color with a step of mold. mold can have a value from 0 to 255.

life_color

Set the color of living (or new born) cells.

death_color

Set the color of dead cells. If mold is set, this is the first color used to represent a dead cell.

mold_color

Set mold color, for definitely dead and moldy cells.

For the syntax of these 3 color options, check the ``Color'' section in the ffmpeg-utils manual.

Examples

*

Read a grid from pattern, and center it on a grid of size 300x300 pixels:

        life=f=pattern:s=300x300

*

Generate a random grid of size 200x200, with a fill ratio of 2/3:

        life=ratio=2/3:s=200x200

*

Specify a custom rule for evolving a randomly generated grid:

        life=rule=S14/B34

*

Full example with slow death effect (mold) using ffplay:

        ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16

allrgb, allyuv, color, haldclutsrc, nullsrc, rgbtestsrc, smptebars, smptehdbars, testsrc, testsrc2, yuvtestsrc

The "allrgb" source returns frames of size 4096x4096 of all rgb colors.

The "allyuv" source returns frames of size 4096x4096 of all yuv colors.

The "color" source provides an uniformly colored input.

The "haldclutsrc" source provides an identity Hald CLUT. See also haldclut filter.

The "nullsrc" source returns unprocessed video frames. It is mainly useful to be employed in analysis / debugging tools, or as the source for filters which ignore the input data.

The "rgbtestsrc" source generates an RGB test pattern useful for detecting RGB vs BGR issues. You should see a red, green and blue stripe from top to bottom.

The "smptebars" source generates a color bars pattern, based on the SMPTE Engineering Guideline EG 1-1990.

The "smptehdbars" source generates a color bars pattern, based on the SMPTE RP 219-2002.

The "testsrc" source generates a test video pattern, showing a color pattern, a scrolling gradient and a timestamp. This is mainly intended for testing purposes.

The "testsrc2" source is similar to testsrc, but supports more pixel formats instead of just "rgb24". This allows using it as an input for other tests without requiring a format conversion.

The "yuvtestsrc" source generates an YUV test pattern. You should see a y, cb and cr stripe from top to bottom.

The sources accept the following parameters:

color, c: Specify the color of the source, only available in the "color" source. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.
level: Specify the level of the Hald CLUT, only available in the "haldclutsrc" source. A level of "N" generates a picture of "N*N*N" by "N*N*N" pixels to be used as identity matrix for 3D lookup tables. Each component is coded on a "1/(N*N)" scale.
size, s: Specify the size of the sourced video. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. The default value is "320x240".
This option is not available with the "haldclutsrc" filter.
rate, r: Specify the frame rate of the sourced video, as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a floating point number or a valid video frame rate abbreviation. The default value is ``25''.
sar: Set the sample aspect ratio of the sourced video.
duration, d: Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax.
If not specified, or the expressed duration is negative, the video is supposed to be generated forever.
decimals, n: Set the number of decimals to show in the timestamp, only available in the "testsrc" source.
The displayed timestamp value will correspond to the original timestamp value multiplied by the power of 10 of the specified value. Default value is 0.

For example the following:

        testsrc=duration=5.3:size=qcif:rate=10

will generate a video with a duration of 5.3 seconds, with size 176x144 and a frame rate of 10 frames per second.

The following graph description will generate a red source with an opacity of 0.2, with size ``qcif'' and a frame rate of 10 frames per second.

        color=c=red@0.2:s=qcif:r=10

If the input content is to be ignored, "nullsrc" can be used. The following command generates noise in the luminance plane by employing the "geq" filter:

        nullsrc=s=256x256, geq=random(1)*255:128:128

Commands

The "color" source supports the following commands:

c, color: Set the color of the created image. Accepts the same syntax of the corresponding color option.

VIDEO SINKS

Below is a description of the currently available video sinks.

buffersink

Buffer video frames, and make them available to the end of the filter graph.

This sink is mainly intended for programmatic use, in particular through the interface defined in libavfilter/buffersink.h or the options system.

It accepts a pointer to an AVBufferSinkContext structure, which defines the incoming buffers' formats, to be passed as the opaque parameter to "avfilter_init_filter" for initialization.

nullsink

Null video sink: do absolutely nothing with the input video. It is mainly useful as a template and for use in analysis / debugging tools.

MULTIMEDIA FILTERS

Below is a description of the currently available multimedia filters.

ahistogram

Convert input audio to a video output, displaying the volume histogram.

The filter accepts the following options:

dmode

Specify how histogram is calculated.

It accepts the following values:

single: Use single histogram for all channels.
separate: Use separate histogram for each channel.

: Default is "single".

rate, r

Set frame rate, expressed as number of frames per second. Default value is ``25''.

size, s

Specify the video size for the output. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default value is "hd720".

scale

Set display scale.

It accepts the following values:

log: logarithmic
sqrt: square root
cbrt: cubic root
lin: linear
rlog: reverse logarithmic

: Default is "log".

ascale

Set amplitude scale.

It accepts the following values:

log: logarithmic
lin: linear

: Default is "log".

acount

Set how much frames to accumulate in histogram. Defauls is 1. Setting this to -1 accumulates all frames.

rheight

Set histogram ratio of window height.

slide

Set sonogram sliding.

It accepts the following values:

replace: replace old rows with new ones.
scroll: scroll from top to bottom.

: Default is "replace".

aphasemeter

Convert input audio to a video output, displaying the audio phase.

The filter accepts the following options:

rate, r: Set the output frame rate. Default value is 25.
size, s: Set the video size for the output. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default value is "800x400".
rc
gc
bc: Specify the red, green, blue contrast. Default values are 2, 7 and 1. Allowed range is "[0, 255]".
mpc: Set color which will be used for drawing median phase. If color is "none" which is default, no median phase value will be drawn.

The filter also exports the frame metadata "lavfi.aphasemeter.phase" which represents mean phase of current audio frame. Value is in range "[-1, 1]". The "-1" means left and right channels are completely out of phase and 1 means channels are in phase.

avectorscope

Convert input audio to a video output, representing the audio vector scope.

The filter is used to measure the difference between channels of stereo audio stream. A monoaural signal, consisting of identical left and right signal, results in straight vertical line. Any stereo separation is visible as a deviation from this line, creating a Lissajous figure. If the straight (or deviation from it) but horizontal line appears this indicates that the left and right channels are out of phase.

The filter accepts the following options:

mode, m

Set the vectorscope mode.

Available values are:

lissajous: Lissajous rotated by 45 degrees.
lissajous_xy: Same as above but not rotated.
polar: Shape resembling half of circle.

: Default value is lissajous.

size, s

Set the video size for the output. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default value is "400x400".

rate, r

Set the output frame rate. Default value is 25.

rc

gc

bc

ac

Specify the red, green, blue and alpha contrast. Default values are 40, 160, 80 and 255. Allowed range is "[0, 255]".

rf

gf

bf

af

Specify the red, green, blue and alpha fade. Default values are 15, 10, 5 and 5. Allowed range is "[0, 255]".

zoom

Set the zoom factor. Default value is 1. Allowed range is "[1, 10]".

draw

Set the vectorscope drawing mode.

Available values are:

dot: Draw dot for each sample.
line: Draw line between previous and current sample.

: Default value is dot.

scale

Specify amplitude scale of audio samples.

Available values are:

lin: Linear.
sqrt: Square root.
cbrt: Cubic root.
log: Logarithmic.

Examples

*

Complete example using ffplay:

        ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
                     [a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]'

bench, abench

Benchmark part of a filtergraph.

The filter accepts the following options:

action

Start or stop a timer.

Available values are:

start: Get the current time, set it as frame metadata (using the key "lavfi.bench.start_time"), and forward the frame to the next filter.
stop: Get the current time and fetch the "lavfi.bench.start_time" metadata from the input frame metadata to get the time difference. Time difference, average, maximum and minimum time (respectively "t", "avg", "max" and "min") are then printed. The timestamps are expressed in seconds.

Examples

*

Benchmark selectivecolor filter:

        bench=start,selectivecolor=reds=-.2 .12 -.49,bench=stop

concat

Concatenate audio and video streams, joining them together one after the other.

The filter works on segments of synchronized video and audio streams. All segments must have the same number of streams of each type, and that will also be the number of streams at output.

The filter accepts the following options:

n: Set the number of segments. Default is 2.
v: Set the number of output video streams, that is also the number of video streams in each segment. Default is 1.
a: Set the number of output audio streams, that is also the number of audio streams in each segment. Default is 0.
unsafe: Activate unsafe mode: do not fail if segments have a different format.

The filter has v+a outputs: first v video outputs, then a audio outputs.

There are nx(v+a) inputs: first the inputs for the first segment, in the same order as the outputs, then the inputs for the second segment, etc.

Related streams do not always have exactly the same duration, for various reasons including codec frame size or sloppy authoring. For that reason, related synchronized streams (e.g. a video and its audio track) should be concatenated at once. The concat filter will use the duration of the longest stream in each segment (except the last one), and if necessary pad shorter audio streams with silence.

For this filter to work correctly, all segments must start at timestamp 0.

All corresponding streams must have the same parameters in all segments; the filtering system will automatically select a common pixel format for video streams, and a common sample format, sample rate and channel layout for audio streams, but other settings, such as resolution, must be converted explicitly by the user.

Different frame rates are acceptable but will result in variable frame rate at output; be sure to configure the output file to handle it.

Examples

*

Concatenate an opening, an episode and an ending, all in bilingual version (video in stream 0, audio in streams 1 and 2):

        ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \
          '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2]
           concat=n=3:v=1:a=2 [v] [a1] [a2]' \
          -map '[v]' -map '[a1]' -map '[a2]' output.mkv

*

Concatenate two parts, handling audio and video separately, using the (a)movie sources, and adjusting the resolution:

        movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ;
        movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ;
        [v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa]

Note that a desync will happen at the stitch if the audio and video streams do not have exactly the same duration in the first file.

drawgraph, adrawgraph

Draw a graph using input video or audio metadata.

It accepts the following parameters:

m1

Set 1st frame metadata key from which metadata values will be used to draw a graph.

fg1

Set 1st foreground color expression.

m2

Set 2nd frame metadata key from which metadata values will be used to draw a graph.

fg2

Set 2nd foreground color expression.

m3

Set 3rd frame metadata key from which metadata values will be used to draw a graph.

fg3

Set 3rd foreground color expression.

m4

Set 4th frame metadata key from which metadata values will be used to draw a graph.

fg4

Set 4th foreground color expression.

min

Set minimal value of metadata value.

max

Set maximal value of metadata value.

bg

Set graph background color. Default is white.

mode

Set graph mode.

Available values for mode is:

bar
dot
line

: Default is "line".

slide

Set slide mode.

Available values for slide is:

frame: Draw new frame when right border is reached.
replace: Replace old columns with new ones.
scroll: Scroll from right to left.
rscroll: Scroll from left to right.
picture: Draw single picture.

: Default is "frame".

size

Set size of graph video. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. The default value is "900x256".

The foreground color expressions can use the following variables:

MIN: Minimal value of metadata value.
MAX: Maximal value of metadata value.
VAL: Current metadata key value.

: The color is defined as 0xAABBGGRR.

Example using metadata from signalstats filter:

        signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255

Example using metadata from ebur128 filter:

        ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5

ebur128

EBU R128 scanner filter. This filter takes an audio stream as input and outputs it unchanged. By default, it logs a message at a frequency of 10Hz with the Momentary loudness (identified by "M"), Short-term loudness ("S"), Integrated loudness ("I") and Loudness Range ("LRA").

The filter also has a video output (see the video option) with a real time graph to observe the loudness evolution. The graphic contains the logged message mentioned above, so it is not printed anymore when this option is set, unless the verbose logging is set. The main graphing area contains the short-term loudness (3 seconds of analysis), and the gauge on the right is for the momentary loudness (400 milliseconds).

More information about the Loudness Recommendation EBU R128 on <http://tech.ebu.ch/loudness>.

The filter accepts the following options:

video

Activate the video output. The audio stream is passed unchanged whether this option is set or no. The video stream will be the first output stream if activated. Default is 0.

size

Set the video size. This option is for video only. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default and minimum resolution is "640x480".

meter

Set the EBU scale meter. Default is 9. Common values are 9 and 18, respectively for EBU scale meter +9 and EBU scale meter +18. Any other integer value between this range is allowed.

metadata

Set metadata injection. If set to 1, the audio input will be segmented into 100ms output frames, each of them containing various loudness information in metadata. All the metadata keys are prefixed with "lavfi.r128.".

Default is 0.

framelog

Force the frame logging level.

Available values are:

info: information logging level
verbose: verbose logging level

: By default, the logging level is set to info. If the video or the metadata options are set, it switches to verbose.

peak

Set peak mode(s).

Available modes can be cumulated (the option is a "flag" type). Possible values are:

none: Disable any peak mode (default).
sample: Enable sample-peak mode.
Simple peak mode looking for the higher sample value. It logs a message for sample-peak (identified by "SPK").
true: Enable true-peak mode.
If enabled, the peak lookup is done on an over-sampled version of the input stream for better peak accuracy. It logs a message for true-peak. (identified by "TPK") and true-peak per frame (identified by "FTPK"). This mode requires a build with "libswresample".

dualmono

Treat mono input files as ``dual mono''. If a mono file is intended for playback on a stereo system, its EBU R128 measurement will be perceptually incorrect. If set to "true", this option will compensate for this effect. Multi-channel input files are not affected by this option.

panlaw

Set a specific pan law to be used for the measurement of dual mono files. This parameter is optional, and has a default value of -3.01dB.

Examples

*

Real-time graph using ffplay, with a EBU scale meter +18:

        ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]"

*

Run an analysis with ffmpeg:

        ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null -

interleave, ainterleave

Temporally interleave frames from several inputs.

"interleave" works with video inputs, "ainterleave" with audio.

These filters read frames from several inputs and send the oldest queued frame to the output.

Input streams must have well defined, monotonically increasing frame timestamp values.

In order to submit one frame to output, these filters need to enqueue at least one frame for each input, so they cannot work in case one input is not yet terminated and will not receive incoming frames.

For example consider the case when one input is a "select" filter which always drops input frames. The "interleave" filter will keep reading from that input, but it will never be able to send new frames to output until the input sends an end-of-stream signal.

Also, depending on inputs synchronization, the filters will drop frames in case one input receives more frames than the other ones, and the queue is already filled.

These filters accept the following options:

nb_inputs, n: Set the number of different inputs, it is 2 by default.

Examples

*

Interleave frames belonging to different streams using ffmpeg:

        ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi

*

Add flickering blur effect:

        select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave

metadata, ametadata

Manipulate frame metadata.

This filter accepts the following options:

mode

Set mode of operation of the filter.

Can be one of the following:

select: If both "value" and "key" is set, select frames which have such metadata. If only "key" is set, select every frame that has such key in metadata.
add: Add new metadata "key" and "value". If key is already available do nothing.
modify: Modify value of already present key.
delete: If "value" is set, delete only keys that have such value. Otherwise, delete key. If "key" is not set, delete all metadata values in the frame.
print: Print key and its value if metadata was found. If "key" is not set print all metadata values available in frame.

key

Set key used with all modes. Must be set for all modes except "print" and "delete".

value

Set metadata value which will be used. This option is mandatory for "modify" and "add" mode.

function

Which function to use when comparing metadata value and "value".

Can be one of following:

same_str: Values are interpreted as strings, returns true if metadata value is same as "value".
starts_with: Values are interpreted as strings, returns true if metadata value starts with the "value" option string.
less: Values are interpreted as floats, returns true if metadata value is less than "value".
equal: Values are interpreted as floats, returns true if "value" is equal with metadata value.
greater: Values are interpreted as floats, returns true if metadata value is greater than "value".
expr: Values are interpreted as floats, returns true if expression from option "expr" evaluates to true.

expr

Set expression which is used when "function" is set to "expr". The expression is evaluated through the eval API and can contain the following constants:

VALUE1: Float representation of "value" from metadata key.
VALUE2: Float representation of "value" as supplied by user in "value" option.
file: If specified in "print" mode, output is written to the named file. Instead of plain filename any writable url can be specified. Filename ``-'' is a shorthand for standard output. If "file" option is not set, output is written to the log with AV_LOG_INFO loglevel.

Examples

*

Print all metadata values for frames with key "lavfi.singnalstats.YDIF" with values between 0 and 1.

        signalstats,metadata=print:key=lavfi.signalstats.YDIF:value=0:function=expr:expr='between(VALUE1,0,1)'

*

Print silencedetect output to file metadata.txt.

        silencedetect,ametadata=mode=print:file=metadata.txt

*

Direct all metadata to a pipe with file descriptor 4.

        metadata=mode=print:file='pipe\:4'

perms, aperms

Set read/write permissions for the output frames.

These filters are mainly aimed at developers to test direct path in the following filter in the filtergraph.

The filters accept the following options:

mode

Select the permissions mode.

It accepts the following values:

none: Do nothing. This is the default.
ro: Set all the output frames read-only.
rw: Set all the output frames directly writable.
toggle: Make the frame read-only if writable, and writable if read-only.
random: Set each output frame read-only or writable randomly.

seed

Set the seed for the random mode, must be an integer included between 0 and "UINT32_MAX". If not specified, or if explicitly set to "-1", the filter will try to use a good random seed on a best effort basis.

Note: in case of auto-inserted filter between the permission filter and the following one, the permission might not be received as expected in that following filter. Inserting a format or aformat filter before the perms/aperms filter can avoid this problem.

realtime, arealtime

Slow down filtering to match real time approximatively.

These filters will pause the filtering for a variable amount of time to match the output rate with the input timestamps. They are similar to the re option to "ffmpeg".

They accept the following options:

limit: Time limit for the pauses. Any pause longer than that will be considered a timestamp discontinuity and reset the timer. Default is 2 seconds.

select, aselect

Select frames to pass in output.

This filter accepts the following options:

expr, e

Set expression, which is evaluated for each input frame.

If the expression is evaluated to zero, the frame is discarded.

If the evaluation result is negative or NaN, the frame is sent to the first output; otherwise it is sent to the output with index "ceil(val)-1", assuming that the input index starts from 0.

For example a value of 1.2 corresponds to the output with index "ceil(1.2)-1 = 2-1 = 1", that is the second output.

outputs, n

Set the number of outputs. The output to which to send the selected frame is based on the result of the evaluation. Default value is 1.

The expression can contain the following constants:

n

The (sequential) number of the filtered frame, starting from 0.

selected_n

The (sequential) number of the selected frame, starting from 0.

prev_selected_n

The sequential number of the last selected frame. It's NAN if undefined.

TB

The timebase of the input timestamps.

pts

The PTS (Presentation TimeStamp) of the filtered video frame, expressed in TB units. It's NAN if undefined.

t

The PTS of the filtered video frame, expressed in seconds. It's NAN if undefined.

prev_pts

The PTS of the previously filtered video frame. It's NAN if undefined.

prev_selected_pts

The PTS of the last previously filtered video frame. It's NAN if undefined.

prev_selected_t

The PTS of the last previously selected video frame. It's NAN if undefined.

start_pts

The PTS of the first video frame in the video. It's NAN if undefined.

start_t

The time of the first video frame in the video. It's NAN if undefined.

pict_type (video only)

The type of the filtered frame. It can assume one of the following values:

I
P
B
S
SI
SP
BI

interlace_type (video only)

The frame interlace type. It can assume one of the following values:

PROGRESSIVE: The frame is progressive (not interlaced).
TOPFIRST: The frame is top-field-first.
BOTTOMFIRST: The frame is bottom-field-first.

consumed_sample_n (audio only)

the number of selected samples before the current frame

samples_n (audio only)

the number of samples in the current frame

sample_rate (audio only)

the input sample rate

key

This is 1 if the filtered frame is a key-frame, 0 otherwise.

pos

the position in the file of the filtered frame, -1 if the information is not available (e.g. for synthetic video)

scene (video only)

value between 0 and 1 to indicate a new scene; a low value reflects a low probability for the current frame to introduce a new scene, while a higher value means the current frame is more likely to be one (see the example below)

concatdec_select

The concat demuxer can select only part of a concat input file by setting an inpoint and an outpoint, but the output packets may not be entirely contained in the selected interval. By using this variable, it is possible to skip frames generated by the concat demuxer which are not exactly contained in the selected interval.

This works by comparing the frame pts against the lavf.concat.start_time and the lavf.concat.duration packet metadata values which are also present in the decoded frames.

The concatdec_select variable is -1 if the frame pts is at least start_time and either the duration metadata is missing or the frame pts is less than start_time + duration, 0 otherwise, and NaN if the start_time metadata is missing.

That basically means that an input frame is selected if its pts is within the interval set by the concat demuxer.

The default value of the select expression is ``1''.

Examples

*

Select all frames in input:

        select

The example above is the same as:

        select=1

*

Skip all frames:

        select=0

*

Select only I-frames:

        select='eq(pict_type\,I)'

*

Select one frame every 100:

        select='not(mod(n\,100))'

*

Select only frames contained in the 10-20 time interval:

        select=between(t\,10\,20)

*

Select only I-frames contained in the 10-20 time interval:

        select=between(t\,10\,20)*eq(pict_type\,I)

*

Select frames with a minimum distance of 10 seconds:

        select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'

*

Use aselect to select only audio frames with samples number > 100:

        aselect='gt(samples_n\,100)'

*

Create a mosaic of the first scenes:

        ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png

Comparing scene against a value between 0.3 and 0.5 is generally a sane choice.

*

Send even and odd frames to separate outputs, and compose them:

        select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h

*

Select useful frames from an ffconcat file which is using inpoints and outpoints but where the source files are not intra frame only.

        ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi

sendcmd, asendcmd

Send commands to filters in the filtergraph.

These filters read commands to be sent to other filters in the filtergraph.

"sendcmd" must be inserted between two video filters, "asendcmd" must be inserted between two audio filters, but apart from that they act the same way.

The specification of commands can be provided in the filter arguments with the commands option, or in a file specified by the filename option.

These filters accept the following options:

commands, c: Set the commands to be read and sent to the other filters.
filename, f: Set the filename of the commands to be read and sent to the other filters.

Commands syntax

A commands description consists of a sequence of interval specifications, comprising a list of commands to be executed when a particular event related to that interval occurs. The occurring event is typically the current frame time entering or leaving a given time interval.

An interval is specified by the following syntax:

        <START>[-<END>] <COMMANDS>;

The time interval is specified by the START and END times. END is optional and defaults to the maximum time.

The current frame time is considered within the specified interval if it is included in the interval [START, END), that is when the time is greater or equal to START and is lesser than END.

COMMANDS consists of a sequence of one or more command specifications, separated by ``,'', relating to that interval. The syntax of a command specification is given by:

        [<FLAGS>] <TARGET> <COMMAND> <ARG>

FLAGS is optional and specifies the type of events relating to the time interval which enable sending the specified command, and must be a non-null sequence of identifier flags separated by ``+'' or ``|'' and enclosed between ``['' and ``]''.

The following flags are recognized:

enter: The command is sent when the current frame timestamp enters the specified interval. In other words, the command is sent when the previous frame timestamp was not in the given interval, and the current is.
leave: The command is sent when the current frame timestamp leaves the specified interval. In other words, the command is sent when the previous frame timestamp was in the given interval, and the current is not.

If FLAGS is not specified, a default value of "[enter]" is assumed.

TARGET specifies the target of the command, usually the name of the filter class or a specific filter instance name.

COMMAND specifies the name of the command for the target filter.

ARG is optional and specifies the optional list of argument for the given COMMAND.

Between one interval specification and another, whitespaces, or sequences of characters starting with "#" until the end of line, are ignored and can be used to annotate comments.

A simplified BNF description of the commands specification syntax follows:

        <COMMAND_FLAG>  ::= "enter" | "leave"
        <COMMAND_FLAGS> ::= <COMMAND_FLAG> [(+|"|")<COMMAND_FLAG>]
        <COMMAND>       ::= ["[" <COMMAND_FLAGS> "]"] <TARGET> <COMMAND> [<ARG>]
        <COMMANDS>      ::= <COMMAND> [,<COMMANDS>]
        <INTERVAL>      ::= <START>[-<END>] <COMMANDS>
        <INTERVALS>     ::= <INTERVAL>[;<INTERVALS>]

Examples

*

Specify audio tempo change at second 4:

        asendcmd=c='4.0 atempo tempo 1.5',atempo

*

Specify a list of drawtext and hue commands in a file.

        # show text in the interval 5-10
        5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
                 [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';
        
        # desaturate the image in the interval 15-20
        15.0-20.0 [enter] hue s 0,
                  [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
                  [leave] hue s 1,
                  [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';
        
        # apply an exponential saturation fade-out effect, starting from time 25
        25 [enter] hue s exp(25-t)

A filtergraph allowing to read and process the above command list stored in a file test.cmd, can be specified with:

        sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue

setpts, asetpts

Change the PTS (presentation timestamp) of the input frames.

"setpts" works on video frames, "asetpts" on audio frames.

This filter accepts the following options:

expr: The expression which is evaluated for each frame to construct its timestamp.

The expression is evaluated through the eval API and can contain the following constants:

FRAME_RATE: frame rate, only defined for constant frame-rate video
PTS: The presentation timestamp in input
N: The count of the input frame for video or the number of consumed samples, not including the current frame for audio, starting from 0.
NB_CONSUMED_SAMPLES: The number of consumed samples, not including the current frame (only audio)
NB_SAMPLES, S: The number of samples in the current frame (only audio)
SAMPLE_RATE, SR: The audio sample rate.
STARTPTS: The PTS of the first frame.
STARTT: the time in seconds of the first frame
INTERLACED: State whether the current frame is interlaced.
T: the time in seconds of the current frame
POS: original position in the file of the frame, or undefined if undefined for the current frame
PREV_INPTS: The previous input PTS.
PREV_INT: previous input time in seconds
PREV_OUTPTS: The previous output PTS.
PREV_OUTT: previous output time in seconds
RTCTIME: The wallclock (RTC) time in microseconds. This is deprecated, use time(0) instead.
RTCSTART: The wallclock (RTC) time at the start of the movie in microseconds.
TB: The timebase of the input timestamps.

Examples

*

Start counting PTS from zero

        setpts=PTS-STARTPTS

*

Apply fast motion effect:

        setpts=0.5*PTS

*

Apply slow motion effect:

        setpts=2.0*PTS

*

Set fixed rate of 25 frames per second:

        setpts=N/(25*TB)

*

Set fixed rate 25 fps with some jitter:

        setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'

*

Apply an offset of 10 seconds to the input PTS:

        setpts=PTS+10/TB

*

Generate timestamps from a ``live source'' and rebase onto the current timebase:

        setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'

*

Generate timestamps by counting samples:

        asetpts=N/SR/TB

settb, asettb

Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase configuration.

It accepts the following parameters:

expr, tb: The expression which is evaluated into the output timebase.

The value for tb is an arithmetic expression representing a rational. The expression can contain the constants ``AVTB'' (the default timebase), ``intb'' (the input timebase) and ``sr'' (the sample rate, audio only). Default value is ``intb''.

Examples

*

Set the timebase to 1/25:

        settb=expr=1/25

*

Set the timebase to 1/10:

        settb=expr=0.1

*

Set the timebase to 1001/1000:

        settb=1+0.001

*

Set the timebase to 2*intb:

        settb=2*intb

*

Set the default timebase value:

        settb=AVTB

showcqt

Convert input audio to a video output representing frequency spectrum logarithmically using Brown-Puckette constant Q transform algorithm with direct frequency domain coefficient calculation (but the transform itself is not really constant Q, instead the Q factor is actually variable/clamped), with musical tone scale, from E0 to D#10.

The filter accepts the following options:

size, s

Specify the video size for the output. It must be even. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default value is "1920x1080".

fps, rate, r

Set the output frame rate. Default value is 25.

bar_h

Set the bargraph height. It must be even. Default value is "-1" which computes the bargraph height automatically.

axis_h

Set the axis height. It must be even. Default value is "-1" which computes the axis height automatically.

sono_h

Set the sonogram height. It must be even. Default value is "-1" which computes the sonogram height automatically.

fullhd

Set the fullhd resolution. This option is deprecated, use size, s instead. Default value is 1.

sono_v, volume

Specify the sonogram volume expression. It can contain variables:

bar_v: the bar_v evaluated expression
frequency, freq, f: the frequency where it is evaluated
timeclamp, tc: the value of timeclamp option

and functions:

a_weighting(f): A-weighting of equal loudness
b_weighting(f): B-weighting of equal loudness
c_weighting(f): C-weighting of equal loudness.

: Default value is 16.

bar_v, volume2

Specify the bargraph volume expression. It can contain variables:

sono_v: the sono_v evaluated expression
frequency, freq, f: the frequency where it is evaluated
timeclamp, tc: the value of timeclamp option

and functions:

a_weighting(f): A-weighting of equal loudness
b_weighting(f): B-weighting of equal loudness
c_weighting(f): C-weighting of equal loudness.

: Default value is "sono_v".

sono_g, gamma

Specify the sonogram gamma. Lower gamma makes the spectrum more contrast, higher gamma makes the spectrum having more range. Default value is 3. Acceptable range is "[1, 7]".

bar_g, gamma2

Specify the bargraph gamma. Default value is 1. Acceptable range is "[1, 7]".

timeclamp, tc

Specify the transform timeclamp. At low frequency, there is trade-off between accuracy in time domain and frequency domain. If timeclamp is lower, event in time domain is represented more accurately (such as fast bass drum), otherwise event in frequency domain is represented more accurately (such as bass guitar). Acceptable range is "[0.1, 1]". Default value is 0.17.

basefreq

Specify the transform base frequency. Default value is 20.01523126408007475, which is frequency 50 cents below E0. Acceptable range is "[10, 100000]".

endfreq

Specify the transform end frequency. Default value is 20495.59681441799654, which is frequency 50 cents above D#10. Acceptable range is "[10, 100000]".

coeffclamp

This option is deprecated and ignored.

tlength

Specify the transform length in time domain. Use this option to control accuracy trade-off between time domain and frequency domain at every frequency sample. It can contain variables:

frequency, freq, f: the frequency where it is evaluated
timeclamp, tc: the value of timeclamp option.

: Default value is "384*tc/(384+tc*f)".

count

Specify the transform count for every video frame. Default value is 6. Acceptable range is "[1, 30]".

fcount

Specify the transform count for every single pixel. Default value is 0, which makes it computed automatically. Acceptable range is "[0, 10]".

fontfile

Specify font file for use with freetype to draw the axis. If not specified, use embedded font. Note that drawing with font file or embedded font is not implemented with custom basefreq and endfreq, use axisfile option instead.

font

Specify fontconfig pattern. This has lower priority than fontfile. The : in the pattern may be replaced by | to avoid unnecessary escaping.

fontcolor

Specify font color expression. This is arithmetic expression that should return integer value 0xRRGGBB. It can contain variables:

frequency, freq, f: the frequency where it is evaluated
timeclamp, tc: the value of timeclamp option

and functions:

midi(f): midi number of frequency f, some midi numbers: E0(16), C1(24), C2(36), A4(69)
r(x), g(x), b(x): red, green, and blue value of intensity x.

: Default value is "st(0, (midi(f)-59.5)/12); st(1, if(between(ld(0),0,1), 0.5-0.5*cos(2*PI*ld(0)), 0)); r(1-ld(1)) + b(ld(1))".

axisfile

Specify image file to draw the axis. This option override fontfile and fontcolor option.

axis, text

Enable/disable drawing text to the axis. If it is set to 0, drawing to the axis is disabled, ignoring fontfile and axisfile option. Default value is 1.

csp

Set colorspace. The accepted values are:

unspecified: Unspecified (default)
bt709: BT.709
fcc: FCC
bt470bg: BT.470BG or BT.601-6 625
smpte170m: SMPTE-170M or BT.601-6 525
smpte240m: SMPTE-240M
bt2020ncl: BT.2020 with non-constant luminance

cscheme

Set spectrogram color scheme. This is list of floating point values with format "left_r|left_g|left_b|right_r|right_g|right_b". The default is "1|0.5|0|0|0.5|1".

Examples

*

Playing audio while showing the spectrum:

        ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'

*

Same as above, but with frame rate 30 fps:

        ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]'

*

Playing at 1280x720:

        ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]'

*

Disable sonogram display:

        sono_h=0

*

A1 and its harmonics: A1, A2, (near)E3, A3:

        ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
                         asplit[a][out1]; [a] showcqt [out0]'

*

Same as above, but with more accuracy in frequency domain:

        ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
                         asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]'

*

Custom volume:

        bar_v=10:sono_v=bar_v*a_weighting(f)

*

Custom gamma, now spectrum is linear to the amplitude.

        bar_g=2:sono_g=2

*

Custom tlength equation:

        tc=0.33:tlength='st(0,0.17); 384*tc / (384 / ld(0) + tc*f /(1-ld(0))) + 384*tc / (tc*f / ld(0) + 384 /(1-ld(0)))'

*

Custom fontcolor and fontfile, C-note is colored green, others are colored blue:

        fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))':fontfile=myfont.ttf

*

Custom font using fontconfig:

        font='Courier New,Monospace,mono|bold'

*

Custom frequency range with custom axis using image file:

        axisfile=myaxis.png:basefreq=40:endfreq=10000

showfreqs

Convert input audio to video output representing the audio power spectrum. Audio amplitude is on Y-axis while frequency is on X-axis.

The filter accepts the following options:

size, s

Specify size of video. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default is "1024x512".

mode

Set display mode. This set how each frequency bin will be represented.

It accepts the following values:

line
bar
dot

: Default is "bar".

ascale

Set amplitude scale.

It accepts the following values:

lin: Linear scale.
sqrt: Square root scale.
cbrt: Cubic root scale.
log: Logarithmic scale.

: Default is "log".

fscale

Set frequency scale.

It accepts the following values:

lin: Linear scale.
log: Logarithmic scale.
rlog: Reverse logarithmic scale.

: Default is "lin".

win_size

Set window size.

It accepts the following values:

w16
w32
w64
w128
w256
w512
w1024
w2048
w4096
w8192
w16384
w32768
w65536

: Default is "w2048"

win_func

Set windowing function.

It accepts the following values:

rect
bartlett
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson

: Default is "hanning".

overlap

Set window overlap. In range "[0, 1]". Default is 1, which means optimal overlap for selected window function will be picked.

averaging

Set time averaging. Setting this to 0 will display current maximal peaks. Default is 1, which means time averaging is disabled.

colors

Specify list of colors separated by space or by '|' which will be used to draw channel frequencies. Unrecognized or missing colors will be replaced by white color.

cmode

Set channel display mode.

It accepts the following values:

combined
separate

: Default is "combined".

minamp

Set minimum amplitude used in "log" amplitude scaler.

showspectrum

Convert input audio to a video output, representing the audio frequency spectrum.

The filter accepts the following options:

size, s

Specify the video size for the output. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default value is "640x512".

slide

Specify how the spectrum should slide along the window.

It accepts the following values:

replace: the samples start again on the left when they reach the right
scroll: the samples scroll from right to left
fullframe: frames are only produced when the samples reach the right
rscroll: the samples scroll from left to right

: Default value is "replace".

mode

Specify display mode.

It accepts the following values:

combined: all channels are displayed in the same row
separate: all channels are displayed in separate rows

: Default value is combined.

color

Specify display color mode.

It accepts the following values:

channel: each channel is displayed in a separate color
intensity: each channel is displayed using the same color scheme
rainbow: each channel is displayed using the rainbow color scheme
moreland: each channel is displayed using the moreland color scheme
nebulae: each channel is displayed using the nebulae color scheme
fire: each channel is displayed using the fire color scheme
fiery: each channel is displayed using the fiery color scheme
fruit: each channel is displayed using the fruit color scheme
cool: each channel is displayed using the cool color scheme

: Default value is channel.

scale

Specify scale used for calculating intensity color values.

It accepts the following values:

lin: linear
sqrt: square root, default
cbrt: cubic root
log: logarithmic
4thrt: 4th root
5thrt: 5th root

: Default value is sqrt.

saturation

Set saturation modifier for displayed colors. Negative values provide alternative color scheme. 0 is no saturation at all. Saturation must be in [-10.0, 10.0] range. Default value is 1.

win_func

Set window function.

It accepts the following values:

rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson

: Default value is "hann".

orientation

Set orientation of time vs frequency axis. Can be "vertical" or "horizontal". Default is "vertical".

overlap

Set ratio of overlap window. Default value is 0. When value is 1 overlap is set to recommended size for specific window function currently used.

gain

Set scale gain for calculating intensity color values. Default value is 1.

data

Set which data to display. Can be "magnitude", default or "phase".

rotation

Set color rotation, must be in [-1.0, 1.0] range. Default value is 0.

The usage is very similar to the showwaves filter; see the examples in that section.

Examples

*

Large window with logarithmic color scaling:

        showspectrum=s=1280x480:scale=log

*

Complete example for a colored and sliding spectrum per channel using ffplay:

        ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
                     [a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'

showspectrumpic

Convert input audio to a single video frame, representing the audio frequency spectrum.

The filter accepts the following options:

size, s

Specify the video size for the output. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default value is "4096x2048".

mode

Specify display mode.

It accepts the following values:

combined: all channels are displayed in the same row
separate: all channels are displayed in separate rows

: Default value is combined.

color

Specify display color mode.

It accepts the following values:

channel: each channel is displayed in a separate color
intensity: each channel is displayed using the same color scheme
rainbow: each channel is displayed using the rainbow color scheme
moreland: each channel is displayed using the moreland color scheme
nebulae: each channel is displayed using the nebulae color scheme
fire: each channel is displayed using the fire color scheme
fiery: each channel is displayed using the fiery color scheme
fruit: each channel is displayed using the fruit color scheme
cool: each channel is displayed using the cool color scheme

: Default value is intensity.

scale

Specify scale used for calculating intensity color values.

It accepts the following values:

lin: linear
sqrt: square root, default
cbrt: cubic root
log: logarithmic
4thrt: 4th root
5thrt: 5th root

: Default value is log.

saturation

Set saturation modifier for displayed colors. Negative values provide alternative color scheme. 0 is no saturation at all. Saturation must be in [-10.0, 10.0] range. Default value is 1.

win_func

Set window function.

It accepts the following values:

rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson

: Default value is "hann".

orientation

Set orientation of time vs frequency axis. Can be "vertical" or "horizontal". Default is "vertical".

gain

Set scale gain for calculating intensity color values. Default value is 1.

legend

Draw time and frequency axes and legends. Default is enabled.

rotation

Set color rotation, must be in [-1.0, 1.0] range. Default value is 0.

Examples

*

Extract an audio spectrogram of a whole audio track in a 1024x1024 picture using ffmpeg:

        ffmpeg -i audio.flac -lavfi showspectrumpic=s=1024x1024 spectrogram.png

showvolume

Convert input audio volume to a video output.

The filter accepts the following options:

rate, r

Set video rate.

b

Set border width, allowed range is [0, 5]. Default is 1.

w

Set channel width, allowed range is [80, 8192]. Default is 400.

h

Set channel height, allowed range is [1, 900]. Default is 20.

f

Set fade, allowed range is [0.001, 1]. Default is 0.95.

c

Set volume color expression.

The expression can use the following variables:

VOLUME: Current max volume of channel in dB.
PEAK: Current peak.
CHANNEL: Current channel number, starting from 0.

t

If set, displays channel names. Default is enabled.

v

If set, displays volume values. Default is enabled.

o

Set orientation, can be "horizontal" or "vertical", default is "horizontal".

s

Set step size, allowed range s [0, 5]. Default is 0, which means step is disabled.

showwaves

Convert input audio to a video output, representing the samples waves.

The filter accepts the following options:

size, s

Specify the video size for the output. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default value is "600x240".

mode

Set display mode.

Available values are:

point: Draw a point for each sample.
line: Draw a vertical line for each sample.
p2p: Draw a point for each sample and a line between them.
cline: Draw a centered vertical line for each sample.

: Default value is "point".

n

Set the number of samples which are printed on the same column. A larger value will decrease the frame rate. Must be a positive integer. This option can be set only if the value for rate is not explicitly specified.

rate, r

Set the (approximate) output frame rate. This is done by setting the option n. Default value is ``25''.

split_channels

Set if channels should be drawn separately or overlap. Default value is 0.

colors

Set colors separated by '|' which are going to be used for drawing of each channel.

scale

Set amplitude scale.

Available values are:

lin: Linear.
log: Logarithmic.
sqrt: Square root.
cbrt: Cubic root.

: Default is linear.

Examples

*

Output the input file audio and the corresponding video representation at the same time:

        amovie=a.mp3,asplit[out0],showwaves[out1]

*

Create a synthetic signal and show it with showwaves, forcing a frame rate of 30 frames per second:

        aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1]

showwavespic

Convert input audio to a single video frame, representing the samples waves.

The filter accepts the following options:

size, s: Specify the video size for the output. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. Default value is "600x240".
split_channels: Set if channels should be drawn separately or overlap. Default value is 0.
colors: Set colors separated by '|' which are going to be used for drawing of each channel.
scale: Set amplitude scale. Can be linear "lin" or logarithmic "log". Default is linear.

Examples

*

Extract a channel split representation of the wave form of a whole audio track in a 1024x800 picture using ffmpeg:

        ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png

sidedata, asidedata

Delete frame side data, or select frames based on it.

This filter accepts the following options:

mode

Set mode of operation of the filter.

Can be one of the following:

select: Select every frame with side data of "type".
delete: Delete side data of "type". If "type" is not set, delete all side data in the frame.

type

Set side data type used with all modes. Must be set for "select" mode. For the list of frame side data types, refer to the "AVFrameSideDataType" enum in libavutil/frame.h. For example, to choose "AV_FRAME_DATA_PANSCAN" side data, you must specify "PANSCAN".

spectrumsynth

Sythesize audio from 2 input video spectrums, first input stream represents magnitude across time and second represents phase across time. The filter will transform from frequency domain as displayed in videos back to time domain as presented in audio output.

This filter is primarily created for reversing processed showspectrum filter outputs, but can synthesize sound from other spectrograms too. But in such case results are going to be poor if the phase data is not available, because in such cases phase data need to be recreated, usually its just recreated from random noise. For best results use gray only output ("channel" color mode in showspectrum filter) and "log" scale for magnitude video and "lin" scale for phase video. To produce phase, for 2nd video, use "data" option. Inputs videos should generally use "fullframe" slide mode as that saves resources needed for decoding video.

The filter accepts the following options:

sample_rate: Specify sample rate of output audio, the sample rate of audio from which spectrum was generated may differ.
channels: Set number of channels represented in input video spectrums.
scale: Set scale which was used when generating magnitude input spectrum. Can be "lin" or "log". Default is "log".
slide: Set slide which was used when generating inputs spectrums. Can be "replace", "scroll", "fullframe" or "rscroll". Default is "fullframe".
win_func: Set window function used for resynthesis.
overlap: Set window overlap. In range "[0, 1]". Default is 1, which means optimal overlap for selected window function will be picked.
orientation: Set orientation of input videos. Can be "vertical" or "horizontal". Default is "vertical".

Examples

*

First create magnitude and phase videos from audio, assuming audio is stereo with 44100 sample rate, then resynthesize videos back to audio with spectrumsynth:

        ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut
        ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut
        ffmpeg -i magnitude.nut -i phase.nut -lavfi spectrumsynth=channels=2:sample_rate=44100:win_func=hann:overlap=0.875:slide=fullframe output.flac

split, asplit

Split input into several identical outputs.

"asplit" works with audio input, "split" with video.

The filter accepts a single parameter which specifies the number of outputs. If unspecified, it defaults to 2.

Examples

*

Create two separate outputs from the same input:

        [in] split [out0][out1]

*

To create 3 or more outputs, you need to specify the number of outputs, like in:

        [in] asplit=3 [out0][out1][out2]

*

Create two separate outputs from the same input, one cropped and one padded:

        [in] split [splitout1][splitout2];
        [splitout1] crop=100:100:0:0    [cropout];
        [splitout2] pad=200:200:100:100 [padout];

*

Create 5 copies of the input audio with ffmpeg:

        ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT

zmq, azmq

Receive commands sent through a libzmq client, and forward them to filters in the filtergraph.

"zmq" and "azmq" work as a pass-through filters. "zmq" must be inserted between two video filters, "azmq" between two audio filters.

To enable these filters you need to install the libzmq library and headers and configure FFmpeg with "--enable-libzmq".

For more information about libzmq see: <http://www.zeromq.org/>

The "zmq" and "azmq" filters work as a libzmq server, which receives messages sent through a network interface defined by the bind_address option.

The received message must be in the form:

        <TARGET> <COMMAND> [<ARG>]

TARGET specifies the target of the command, usually the name of the filter class or a specific filter instance name.

COMMAND specifies the name of the command for the target filter.

ARG is optional and specifies the optional argument list for the given COMMAND.

Upon reception, the message is processed and the corresponding command is injected into the filtergraph. Depending on the result, the filter will send a reply to the client, adopting the format:

        <ERROR_CODE> <ERROR_REASON>
        <MESSAGE>

MESSAGE is optional.

Examples

Look at tools/zmqsend for an example of a zmq client which can be used to send commands processed by these filters.

Consider the following filtergraph generated by ffplay

        ffplay -dumpgraph 1 -f lavfi "
        color=s=100x100:c=red  [l];
        color=s=100x100:c=blue [r];
        nullsrc=s=200x100, zmq [bg];
        [bg][l]   overlay      [bg+l];
        [bg+l][r] overlay=x=100 "

To change the color of the left side of the video, the following command can be used:

        echo Parsed_color_0 c yellow | tools/zmqsend

To change the right side:

        echo Parsed_color_1 c pink | tools/zmqsend

MULTIMEDIA SOURCES

Below is a description of the currently available multimedia sources.

amovie

This is the same as movie source, except it selects an audio stream by default.

movie

Read audio and/or video stream(s) from a movie container.

It accepts the following parameters:

filename: The name of the resource to read (not necessarily a file; it can also be a device or a stream accessed through some protocol).
format_name, f: Specifies the format assumed for the movie to read, and can be either the name of a container or an input device. If not specified, the format is guessed from movie_name or by probing.
seek_point, sp: Specifies the seek point in seconds. The frames will be output starting from this seek point. The parameter is evaluated with "av_strtod", so the numerical value may be suffixed by an IS postfix. The default value is ``0''.
streams, s: Specifies the streams to read. Several streams can be specified, separated by ``+''. The source will then have as many outputs, in the same order. The syntax is explained in the ``Stream specifiers'' section in the ffmpeg manual. Two special names, ``dv'' and ``da'' specify respectively the default (best suited) video and audio stream. Default is ``dv'', or ``da'' if the filter is called as ``amovie''.
stream_index, si: Specifies the index of the video stream to read. If the value is -1, the most suitable video stream will be automatically selected. The default value is ``-1''. Deprecated. If the filter is called ``amovie'', it will select audio instead of video.
loop: Specifies how many times to read the stream in sequence. If the value is less than 1, the stream will be read again and again. Default value is ``1''.
Note that when the movie is looped the source timestamps are not changed, so it will generate non monotonically increasing timestamps.
discontinuity: Specifies the time difference between frames above which the point is considered a timestamp discontinuity which is removed by adjusting the later timestamps.

It allows overlaying a second video on top of the main input of a filtergraph, as shown in this graph:

        input -----------> deltapts0 --> overlay --> output
                                            ^
                                            |
        movie --> scale--> deltapts1 -------+

Examples

*

Skip 3.2 seconds from the start of the AVI file in.avi, and overlay it on top of the input labelled ``in'':

        movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over];
        [in] setpts=PTS-STARTPTS [main];
        [main][over] overlay=16:16 [out]

*

Read from a video4linux2 device, and overlay it on top of the input labelled ``in'':

        movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over];
        [in] setpts=PTS-STARTPTS [main];
        [main][over] overlay=16:16 [out]

*

Read the first video stream and the audio stream with id 0x81 from dvd.vob; the video is connected to the pad named ``video'' and the audio is connected to the pad named ``audio'':

        movie=dvd.vob:s=v:0+#0x81 [video] [audio]

Commands

Both movie and amovie support the following commands:

seek

Perform seek using ``av_seek_frame''. The syntax is: seek stream_index|timestamp|flags

*: stream_index: If stream_index is -1, a default stream is selected, and timestamp is automatically converted from AV_TIME_BASE units to the stream specific time_base.
*: timestamp: Timestamp in AVStream.time_base units or, if no stream is specified, in AV_TIME_BASE units.
*: flags: Flags which select direction and seeking mode.

get_duration

Get movie duration in AV_TIME_BASE units.

AUTHORS

The FFmpeg developers.

For details about the authorship, see the Git history of the project (git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in the FFmpeg source directory, or browsing the online repository at <http://source.ffmpeg.org>.

Maintainers for the specific components are listed in the file MAINTAINERS in the source code tree.