ffmpeg [global_options] {[input_file_options] -i ‘input_url’} ... {[output_file_options] ‘output_url’} ...
ffmpeg is a very fast video and audio converter that can also grab from
a live audio/video source. It can also convert between arbitrary sample
rates and resize video on the fly with a high quality polyphase filter.
ffmpeg reads from an arbitrary number of input "files" (which can be regular
files, pipes, network streams, grabbing devices, etc.), specified by the
-i option, and writes to an arbitrary number of output "files", which are
specified by a plain output url. Anything found on the command line which
cannot be interpreted as an option is considered to be an output url.
Each input or output url can, in principle, contain any number of streams of
different types (video/audio/subtitle/attachment/data). The allowed number and/or
types of streams may be limited by the container format. Selecting which
streams from which inputs will go into which output is either done automatically
or with the -map option (see the Stream selection chapter).
To refer to input files in options, you must use their indices (0-based). E.g.
the first input file is 0, the second is 1, etc. Similarly, streams
within a file are referred to by their indices. E.g. 2:3 refers to the
fourth stream in the third input file. Also see the Stream specifiers chapter.
As a general rule, options are applied to the next specified file. Therefore, order is important, and you can have the same option on the command line multiple times. Each occurrence is then applied to the next input or output file. Exceptions from this rule are the global options (e.g. verbosity level), which should be specified first.
Do not mix input and output files – first specify all input files, then all output files. Also do not mix options which belong to different files. All options apply ONLY to the next input or output file and are reset between files.
ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi
ffmpeg -i input.avi -r 24 output.avi
ffmpeg -r 1 -i input.m2v -r 24 output.avi
The format option may be needed for raw input files.
The transcoding process in ffmpeg for each output can be described by
the following diagram:
_______ ______________
| | | |
| input | demuxer | encoded data | decoder
| file | ---------> | packets | -----+
|_______| |______________| |
v
_________
| |
| decoded |
| frames |
|_________|
________ ______________ |
| | | | |
| output | <-------- | encoded data | <----+
| file | muxer | packets | encoder
|________| |______________|
ffmpeg calls the libavformat library (containing demuxers) to read
input files and get packets containing encoded data from them. When there are
multiple input files, ffmpeg tries to keep them synchronized by
tracking lowest timestamp on any active input stream.
Encoded packets are then passed to the decoder (unless streamcopy is selected for the stream, see further for a description). The decoder produces uncompressed frames (raw video/PCM audio/...) which can be processed further by filtering (see next section). After filtering, the frames are passed to the encoder, which encodes them and outputs encoded packets. Finally those are passed to the muxer, which writes the encoded packets to the output file.
Before encoding, ffmpeg can process raw audio and video frames using
filters from the libavfilter library. Several chained filters form a filter
graph. ffmpeg distinguishes between two types of filtergraphs:
simple and complex.
Simple filtergraphs are those that have exactly one input and output, both of the same type. In the above diagram they can be represented by simply inserting an additional step between decoding and encoding:
_________ ______________
| | | |
| decoded | | encoded data |
| frames |\ _ | packets |
|_________| \ /||______________|
\ __________ /
simple _\|| | / encoder
filtergraph | filtered |/
| frames |
|__________|
Simple filtergraphs are configured with the per-stream ‘-filter’ option (with ‘-vf’ and ‘-af’ aliases for video and audio respectively). A simple filtergraph for video can look for example like this:
_______ _____________ _______ ________ | | | | | | | | | input | ---> | deinterlace | ---> | scale | ---> | output | |_______| |_____________| |_______| |________|
Note that some filters change frame properties but not frame contents. E.g. the
fps filter in the example above changes number of frames, but does not
touch the frame contents. Another example is the setpts filter, which
only sets timestamps and otherwise passes the frames unchanged.
Complex filtergraphs are those which cannot be described as simply a linear processing chain applied to one stream. This is the case, for example, when the graph has more than one input and/or output, or when output stream type is different from input. They can be represented with the following diagram:
_________
| |
| input 0 |\ __________
|_________| \ | |
\ _________ /| output 0 |
\ | | / |__________|
_________ \| complex | /
| | | |/
| input 1 |---->| filter |\
|_________| | | \ __________
/| graph | \ | |
/ | | \| output 1 |
_________ / |_________| |__________|
| | /
| input 2 |/
|_________|
Complex filtergraphs are configured with the ‘-filter_complex’ option. Note that this option is global, since a complex filtergraph, by its nature, cannot be unambiguously associated with a single stream or file.
The ‘-lavfi’ option is equivalent to ‘-filter_complex’.
A trivial example of a complex filtergraph is the overlay filter, which
has two video inputs and one video output, containing one video overlaid on top
of the other. Its audio counterpart is the amix filter.
Stream copy is a mode selected by supplying the copy parameter to the
‘-codec’ option. It makes ffmpeg omit the decoding and encoding
step for the specified stream, so it does only demuxing and muxing. It is useful
for changing the container format or modifying container-level metadata. The
diagram above will, in this case, simplify to this:
_______ ______________ ________ | | | | | | | input | demuxer | encoded data | muxer | output | | file | ---------> | packets | -------> | file | |_______| |______________| |________|
Since there is no decoding or encoding, it is very fast and there is no quality loss. However, it might not work in some cases because of many factors. Applying filters is obviously also impossible, since filters work on uncompressed data.
By default, ffmpeg includes only one stream of each type (video, audio, subtitle)
present in the input files and adds them to each output file. It picks the
"best" of each based upon the following criteria: for video, it is the stream
with the highest resolution, for audio, it is the stream with the most channels, for
subtitles, it is the first subtitle stream. In the case where several streams of
the same type rate equally, the stream with the lowest index is chosen.
You can disable some of those defaults by using the -vn/-an/-sn/-dn options. For
full manual control, use the -map option, which disables the defaults just
described.
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.
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:
Matches the stream with this index. E.g. -threads:1 4 would set the
thread count for the second stream to 4.
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:m:key[:value] In first version, 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. In the second version, stream_type is one of following: ’v’ for video, ’a’ for audio, ’s’ for subtitle, ’d’ for data. If stream_index is also given, then it matches stream number stream_index of this type in the program with the id program_id. Otherwise, if only stream_type is given, it matches all streams of this type in the program with the id program_id. In the third version matches streams in the program with the id program_id with the metadata tag key having the specified value. If value is not given, matches streams that contain the given tag with any value.
Match the stream by stream id (e.g. PID in MPEG-TS container).
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.
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.
These options are shared amongst the ff* tools.
Show license.
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:
Print advanced tool options in addition to the basic tool options.
Print complete list of options, including shared and private options for encoders, decoders, demuxers, muxers, filters, etc.
Print detailed information about the decoder named decoder_name. Use the ‘-decoders’ option to get a list of all decoders.
Print detailed information about the encoder named encoder_name. Use the ‘-encoders’ option to get a list of all encoders.
Print detailed information about the demuxer named demuxer_name. Use the ‘-formats’ option to get a list of all demuxers and muxers.
Print detailed information about the muxer named muxer_name. Use the ‘-formats’ option to get a list of all muxers and demuxers.
Print detailed information about the filter name filter_name. Use the ‘-filters’ option to get a list of all filters.
Show version.
Show available formats (including devices).
Show available demuxers.
Show available muxers.
Show available devices.
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.
Show available decoders.
Show all available encoders.
Show available bitstream filters.
Show available protocols.
Show available libavfilter filters.
Show available pixel formats.
Show available sample formats.
Show channel names and standard channel layouts.
Show recognized color names.
Show autodetected sources of the input device. Some devices may provide system-dependent source names that cannot be autodetected. The returned list cannot be assumed to be always complete.
ffmpeg -sources pulse,server=192.168.0.4
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
Set logging level and flags used by the library.
The optional flags prefix can consist of the following values:
Indicates that repeated log output should not be compressed to the first line and the "Last message repeated n times" line will be omitted.
Indicates that log output should add a [level] prefix to each message
line. This can be used as an alternative to log coloring, e.g. when dumping the
log to file.
Flags can also be used alone by adding a ’+’/’-’ prefix to set/reset a single flag without affecting other flags or changing loglevel. When setting both flags and loglevel, a ’+’ separator is expected between the last flags value and before loglevel.
loglevel is a string or a number containing one of the following values:
Show nothing at all; be silent.
Only show fatal errors which could lead the process to crash, such as an assertion failure. This is not currently used for anything.
Only show fatal errors. These are errors after which the process absolutely cannot continue.
Show all errors, including ones which can be recovered from.
Show all warnings and errors. Any message related to possibly incorrect or unexpected events will be shown.
Show informative messages during processing. This is in addition to warnings and errors. This is the default value.
Same as info, except more verbose.
Show everything, including debugging information.
For example to enable repeated log output, add the level prefix, and set
loglevel to verbose:
ffmpeg -loglevel repeat+level+verbose -i input output
Another example that enables repeated log output without affecting current
state of level prefix flag or loglevel:
ffmpeg [...] -loglevel +repeat
By default the program logs to stderr. If coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
AV_LOG_FORCE_NOCOLOR or 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.
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:
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 %
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.
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.
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:
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:
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.
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.
Force input or output file format. The format is normally auto detected for input files and guessed from the file extension for output files, so this option is not needed in most cases.
input file url
Overwrite output files without asking.
Do not overwrite output files, and exit immediately if a specified output file already exists.
Set number of times input stream shall be looped. Loop 0 means no loop, loop -1 means infinite loop.
Select an encoder (when used before an output file) or a decoder (when used
before an input file) for one or more streams. codec is the name of a
decoder/encoder or a special value copy (output only) to indicate that
the stream is not to be re-encoded.
For example
ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT
encodes all video streams with libx264 and copies all audio streams.
For each stream, the last matching c option is applied, so
ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT
will copy all the streams except the second video, which will be encoded with libx264, and the 138th audio, which will be encoded with libvorbis.
When used as an input option (before -i), limit the duration of
data read from the input file.
When used as an output option (before an output url), stop writing the output after its duration reaches duration.
duration must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.
-to and -t are mutually exclusive and -t has priority.
Stop writing the output or reading the input at position. position must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.
-to and -t are mutually exclusive and -t has priority.
Set the file size limit, expressed in bytes. No further chunk of bytes is written after the limit is exceeded. The size of the output file is slightly more than the requested file size.
When used as an input option (before -i), seeks in this input file to
position. Note that in most formats it is not possible to seek exactly,
so ffmpeg will seek to the closest seek point before position.
When transcoding and ‘-accurate_seek’ is enabled (the default), this
extra segment between the seek point and position will be decoded and
discarded. When doing stream copy or when ‘-noaccurate_seek’ is used, it
will be preserved.
When used as an output option (before an output url), decodes but discards input until the timestamps reach position.
position must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.
Like the -ss option but relative to the "end of file". That is negative
values are earlier in the file, 0 is at EOF.
Set the input time offset.
offset must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.
The offset is added to the timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by the time duration specified in offset.
Set the recording timestamp in the container.
date must be a date specification, see (ffmpeg-utils)the Date section in the ffmpeg-utils(1) manual.
Set a metadata key/value pair.
An optional metadata_specifier may be given to set metadata
on streams, chapters or programs. See -map_metadata
documentation for details.
This option overrides metadata set with -map_metadata. It is
also possible to delete metadata by using an empty value.
For example, for setting the title in the output file:
ffmpeg -i in.avi -metadata title="my title" out.flv
To set the language of the first audio stream:
ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
Sets the disposition for a stream.
This option overrides the disposition copied from the input stream. It is also possible to delete the disposition by setting it to 0.
The following dispositions are recognized:
For example, to make the second audio stream the default stream:
ffmpeg -i in.mkv -disposition:a:1 default out.mkv
To make the second subtitle stream the default stream and remove the default disposition from the first subtitle stream:
ffmpeg -i INPUT -disposition:s:0 0 -disposition:s:1 default OUTPUT
Creates a program with the specified title, program_num and adds the specified stream(s) to it.
Specify target file type (vcd, svcd, dvd, dv,
dv50). type may be prefixed with pal-, ntsc- or
film- to use the corresponding standard. All the format options
(bitrate, codecs, buffer sizes) are then set automatically. You can just type:
ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg
Nevertheless you can specify additional options as long as you know they do not conflict with the standard, as in:
ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
Disable data recording. For full manual control see the -map
option.
Set the number of data frames to output. This is an obsolete alias for
-frames:d, which you should use instead.
Stop writing to the stream after framecount frames.
Use fixed quality scale (VBR). The meaning of q/qscale is codec-dependent. If qscale is used without a stream_specifier then it applies only to the video stream, this is to maintain compatibility with previous behavior and as specifying the same codec specific value to 2 different codecs that is audio and video generally is not what is intended when no stream_specifier is used.
Create the filtergraph specified by filtergraph and use it to filter the stream.
filtergraph is a description of the filtergraph to apply to
the stream, and must have a single input and a single output of the
same type of the stream. In the filtergraph, the input is associated
to the label in, and the output to the label out. See
the ffmpeg-filters manual for more information about the filtergraph
syntax.
See the -filter_complex option if you want to create filtergraphs with multiple inputs and/or outputs.
This option is similar to ‘-filter’, the only difference is that its argument is the name of the file from which a filtergraph description is to be read.
Defines how many threads are used to process a filter pipeline. Each pipeline will produce a thread pool with this many threads available for parallel processing. The default is the number of available CPUs.
Specify the preset for matching stream(s).
Print encoding progress/statistics. It is on by default, to explicitly
disable it you need to specify -nostats.
Send program-friendly progress information to url.
Progress information is written approximately every second and at the end of the encoding process. It is made of "key=value" lines. key consists of only alphanumeric characters. The last key of a sequence of progress information is always "progress".
Enable interaction on standard input. On by default unless standard input is
used as an input. To explicitly disable interaction you need to specify
-nostdin.
Disabling interaction on standard input is useful, for example, if
ffmpeg is in the background process group. Roughly the same result can
be achieved with ffmpeg ... < /dev/null but it requires a
shell.
Print timestamp information. It is off by default. This option is mostly useful for testing and debugging purposes, and the output format may change from one version to another, so it should not be employed by portable scripts.
See also the option -fdebug ts.
Add an attachment to the output file. This is supported by a few formats
like Matroska for e.g. fonts used in rendering subtitles. Attachments
are implemented as a specific type of stream, so this option will add
a new stream to the file. It is then possible to use per-stream options
on this stream in the usual way. Attachment streams created with this
option will be created after all the other streams (i.e. those created
with -map or automatic mappings).
Note that for Matroska you also have to set the mimetype metadata tag:
ffmpeg -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv
(assuming that the attachment stream will be third in the output file).
Extract the matching attachment stream into a file named filename. If
filename is empty, then the value of the filename metadata tag
will be used.
E.g. to extract the first attachment to a file named ’out.ttf’:
ffmpeg -dump_attachment:t:0 out.ttf -i INPUT
To extract all attachments to files determined by the filename tag:
ffmpeg -dump_attachment:t "" -i INPUT
Technical note – attachments are implemented as codec extradata, so this option can actually be used to extract extradata from any stream, not just attachments.
Disable automatically rotating video based on file metadata.
Set the number of video frames to output. This is an obsolete alias for
-frames:v, which you should use instead.
Set frame rate (Hz value, fraction or abbreviation).
As an input option, ignore any timestamps stored in the file and instead generate timestamps assuming constant frame rate fps. This is not the same as the ‘-framerate’ option used for some input formats like image2 or v4l2 (it used to be the same in older versions of FFmpeg). If in doubt use ‘-framerate’ instead of the input option ‘-r’.
As an output option, duplicate or drop input frames to achieve constant output frame rate fps.
Set frame size.
As an input option, this is a shortcut for the ‘video_size’ private option, recognized by some demuxers for which the frame size is either not stored in the file or is configurable – e.g. raw video or video grabbers.
As an output option, this inserts the scale video filter to the
end of the corresponding filtergraph. Please use the scale filter
directly to insert it at the beginning or some other place.
The format is ‘wxh’ (default - same as source).
Set the video display aspect ratio specified by aspect.
aspect can be a floating point number string, or a string of the form num:den, where num and den are the numerator and denominator of the aspect ratio. For example "4:3", "16:9", "1.3333", and "1.7777" are valid argument values.
If used together with ‘-vcodec copy’, it will affect the aspect ratio stored at container level, but not the aspect ratio stored in encoded frames, if it exists.
Disable video recording. For full manual control see the -map
option.
Set the video codec. This is an alias for -codec:v.
Select the pass number (1 or 2). It is used to do two-pass video encoding. The statistics of the video are recorded in the first pass into a log file (see also the option -passlogfile), and in the second pass that log file is used to generate the video at the exact requested bitrate. On pass 1, you may just deactivate audio and set output to null, examples for Windows and Unix:
ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null
Set two-pass log file name prefix to prefix, the default file name prefix is “ffmpeg2pass”. The complete file name will be ‘PREFIX-N.log’, where N is a number specific to the output stream
Create the filtergraph specified by filtergraph and use it to filter the stream.
This is an alias for -filter:v, see the -filter option.
Set pixel format. Use -pix_fmts to show all the supported
pixel formats.
If the selected pixel format can not be selected, ffmpeg will print a
warning and select the best pixel format supported by the encoder.
If pix_fmt is prefixed by a +, ffmpeg will exit with an error
if the requested pixel format can not be selected, and automatic conversions
inside filtergraphs are disabled.
If pix_fmt is a single +, ffmpeg selects the same pixel format
as the input (or graph output) and automatic conversions are disabled.
Set SwScaler flags.
Discard threshold.
Rate control override for specific intervals, formatted as "int,int,int" list separated with slashes. Two first values are the beginning and end frame numbers, last one is quantizer to use if positive, or quality factor if negative.
Force interlacing support in encoder (MPEG-2 and MPEG-4 only). Use this option if your input file is interlaced and you want to keep the interlaced format for minimum losses. The alternative is to deinterlace the input stream with ‘-deinterlace’, but deinterlacing introduces losses.
Calculate PSNR of compressed frames.
Dump video coding statistics to ‘vstats_HHMMSS.log’.
Dump video coding statistics to file.
Specifies which version of the vstats format to use. Default is 2.
version = 1 :
frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s
version > 1:
out= %2d st= %2d frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s
top=1/bottom=0/auto=-1 field first
Intra_dc_precision.
Force video tag/fourcc. This is an alias for -tag:v.
Show QP histogram
Deprecated see -bsf
Force key frames at the specified timestamps, more precisely at the first frames after each specified time.
If the argument is prefixed with expr:, the string expr
is interpreted like an expression and is evaluated for each frame. A
key frame is forced in case the evaluation is non-zero.
If one of the times is "chapters[delta]", it is expanded into
the time of the beginning of all chapters in the file, shifted by
delta, expressed as a time in seconds.
This option can be useful to ensure that a seek point is present at a
chapter mark or any other designated place in the output file.
For example, to insert a key frame at 5 minutes, plus key frames 0.1 second before the beginning of every chapter:
-force_key_frames 0:05:00,chapters-0.1
The expression in expr can contain the following constants:
the number of current processed frame, starting from 0
the number of forced frames
the number of the previous forced frame, it is NAN when no
keyframe was forced yet
the time of the previous forced frame, it is NAN when no
keyframe was forced yet
the time of the current processed frame
For example to force a key frame every 5 seconds, you can specify:
-force_key_frames expr:gte(t,n_forced*5)
To force a key frame 5 seconds after the time of the last forced one, starting from second 13:
-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))
Note that forcing too many keyframes is very harmful for the lookahead algorithms of certain encoders: using fixed-GOP options or similar would be more efficient.
When doing stream copy, copy also non-key frames found at the beginning.
Initialise a new hardware device of type type called name, using the given device parameters. If no name is specified it will receive a default name of the form "type%d".
The meaning of device and the following arguments depends on the device type:
device is the number of the CUDA device.
device is the number of the Direct3D 9 display adapter.
device is either an X11 display name or a DRM render node. If not specified, it will attempt to open the default X11 display ($DISPLAY) and then the first DRM render node (/dev/dri/renderD128).
device is an X11 display name. If not specified, it will attempt to open the default X11 display ($DISPLAY).
device selects a value in ‘MFX_IMPL_*’. Allowed values are:
If not specified, ‘auto_any’ is used. (Note that it may be easier to achieve the desired result for QSV by creating the platform-appropriate subdevice (‘dxva2’ or ‘vaapi’) and then deriving a QSV device from that.)
device selects the platform and device as platform_index.device_index.
The set of devices can also be filtered using the key-value pairs to find only devices matching particular platform or device strings.
The strings usable as filters are:
The indices and filters must together uniquely select a device.
Examples:
Choose the second device on the first platform.
Choose the device with a name containing the string Foo9000.
Choose the GPU device on the second platform supporting the cl_khr_fp16 extension.
Initialise a new hardware device of type type called name, deriving it from the existing device with the name source.
List all hardware device types supported in this build of ffmpeg.
Pass the hardware device called name to all filters in any filter graph.
This can be used to set the device to upload to with the hwupload filter,
or the device to map to with the hwmap filter. Other filters may also
make use of this parameter when they require a hardware device. Note that this
is typically only required when the input is not already in hardware frames -
when it is, filters will derive the device they require from the context of the
frames they receive as input.
This is a global setting, so all filters will receive the same device.
Use hardware acceleration to decode the matching stream(s). The allowed values of hwaccel are:
Do not use any hardware acceleration (the default).
Automatically select the hardware acceleration method.
Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
Use VAAPI (Video Acceleration API) hardware acceleration.
Use the Intel QuickSync Video acceleration for video transcoding.
Unlike most other values, this option does not enable accelerated decoding (that is used automatically whenever a qsv decoder is selected), but accelerated transcoding, without copying the frames into the system memory.
For it to work, both the decoder and the encoder must support QSV acceleration and no filters must be used.
This option has no effect if the selected hwaccel is not available or not supported by the chosen decoder.
Note that most acceleration methods are intended for playback and will not be
faster than software decoding on modern CPUs. Additionally, ffmpeg
will usually need to copy the decoded frames from the GPU memory into the system
memory, resulting in further performance loss. This option is thus mainly
useful for testing.
Select a device to use for hardware acceleration.
This option only makes sense when the ‘-hwaccel’ option is also specified. It can either refer to an existing device created with ‘-init_hw_device’ by name, or it can create a new device as if ‘-init_hw_device’ type:hwaccel_device were called immediately before.
List all hardware acceleration methods supported in this build of ffmpeg.
Set the number of audio frames to output. This is an obsolete alias for
-frames:a, which you should use instead.
Set the audio sampling frequency. For output streams it is set by default to the frequency of the corresponding input stream. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options.
Set the audio quality (codec-specific, VBR). This is an alias for -q:a.
Set the number of audio channels. For output streams it is set by default to the number of input audio channels. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options.
Disable audio recording. For full manual control see the -map
option.
Set the audio codec. This is an alias for -codec:a.
Set the audio sample format. Use -sample_fmts to get a list
of supported sample formats.
Create the filtergraph specified by filtergraph and use it to filter the stream.
This is an alias for -filter:a, see the -filter option.
Force audio tag/fourcc. This is an alias for -tag:a.
Deprecated, see -bsf
If some input channel layout is not known, try to guess only if it
corresponds to at most the specified number of channels. For example, 2
tells to ffmpeg to recognize 1 channel as mono and 2 channels as
stereo but not 6 channels as 5.1. The default is to always try to guess. Use
0 to disable all guessing.
Set the subtitle codec. This is an alias for -codec:s.
Disable subtitle recording. For full manual control see the -map
option.
Deprecated, see -bsf
Fix subtitles durations. For each subtitle, wait for the next packet in the same stream and adjust the duration of the first to avoid overlap. This is necessary with some subtitles codecs, especially DVB subtitles, because the duration in the original packet is only a rough estimate and the end is actually marked by an empty subtitle frame. Failing to use this option when necessary can result in exaggerated durations or muxing failures due to non-monotonic timestamps.
Note that this option will delay the output of all data until the next subtitle packet is decoded: it may increase memory consumption and latency a lot.
Set the size of the canvas used to render subtitles.
Designate one or more input streams as a source for the output file. Each input stream is identified by the input file index input_file_id and the input stream index input_stream_id within the input file. Both indices start at 0. If specified, sync_file_id:stream_specifier sets which input stream is used as a presentation sync reference.
The first -map option on the command line specifies the
source for output stream 0, the second -map option specifies
the source for output stream 1, etc.
A - character before the stream identifier creates a "negative" mapping.
It disables matching streams from already created mappings.
A trailing ? after the stream index will allow the map to be
optional: if the map matches no streams the map will be ignored instead
of failing. Note the map will still fail if an invalid input file index
is used; such as if the map refers to a non-existent input.
An alternative [linklabel] form will map outputs from complex filter graphs (see the ‘-filter_complex’ option) to the output file. linklabel must correspond to a defined output link label in the graph.
For example, to map ALL streams from the first input file to output
ffmpeg -i INPUT -map 0 output
For example, if you have two audio streams in the first input file,
these streams are identified by "0:0" and "0:1". You can use
-map to select which streams to place in an output file. For
example:
ffmpeg -i INPUT -map 0:1 out.wav
will map the input stream in ‘INPUT’ identified by "0:1" to the (single) output stream in ‘out.wav’.
For example, to select the stream with index 2 from input file ‘a.mov’ (specified by the identifier "0:2"), and stream with index 6 from input ‘b.mov’ (specified by the identifier "1:6"), and copy them to the output file ‘out.mov’:
ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov
To select all video and the third audio stream from an input file:
ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT
To map all the streams except the second audio, use negative mappings
ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
To map the video and audio streams from the first input, and using the
trailing ?, ignore the audio mapping if no audio streams exist in
the first input:
ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT
To pick the English audio stream:
ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
Note that using this option disables the default mappings for this output file.
Ignore input streams with unknown type instead of failing if copying such streams is attempted.
Allow input streams with unknown type to be copied instead of failing if copying such streams is attempted.
Map an audio channel from a given input to an output. If output_file_id.stream_specifier is not set, the audio channel will be mapped on all the audio streams.
Using "-1" instead of input_file_id.stream_specifier.channel_id will map a muted channel.
A trailing ? will allow the map_channel to be
optional: if the map_channel matches no channel the map_channel will be ignored instead
of failing.
For example, assuming INPUT is a stereo audio file, you can switch the two audio channels with the following command:
ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT
If you want to mute the first channel and keep the second:
ffmpeg -i INPUT -map_channel -1 -map_channel 0.0.1 OUTPUT
The order of the "-map_channel" option specifies the order of the channels in the output stream. The output channel layout is guessed from the number of channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac" in combination of "-map_channel" makes the channel gain levels to be updated if input and output channel layouts don’t match (for instance two "-map_channel" options and "-ac 6").
You can also extract each channel of an input to specific outputs; the following command extracts two channels of the INPUT audio stream (file 0, stream 0) to the respective OUTPUT_CH0 and OUTPUT_CH1 outputs:
ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1
The following example splits the channels of a stereo input into two separate streams, which are put into the same output file:
ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg
Note that currently each output stream can only contain channels from a single input stream; you can’t for example use "-map_channel" to pick multiple input audio channels contained in different streams (from the same or different files) and merge them into a single output stream. It is therefore not currently possible, for example, to turn two separate mono streams into a single stereo stream. However splitting a stereo stream into two single channel mono streams is possible.
If you need this feature, a possible workaround is to use the amerge filter. For example, if you need to merge a media (here ‘input.mkv’) with 2 mono audio streams into one single stereo channel audio stream (and keep the video stream), you can use the following command:
ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv
To map the first two audio channels from the first input, and using the
trailing ?, ignore the audio channel mapping if the first input is
mono instead of stereo:
ffmpeg -i INPUT -map_channel 0.0.0 -map_channel 0.0.1? OUTPUT
Set metadata information of the next output file from infile. Note that those are file indices (zero-based), not filenames. Optional metadata_spec_in/out parameters specify, which metadata to copy. A metadata specifier can have the following forms:
global metadata, i.e. metadata that applies to the whole file
per-stream metadata. stream_spec is a stream specifier as described in the Stream specifiers chapter. In an input metadata specifier, the first matching stream is copied from. In an output metadata specifier, all matching streams are copied to.
per-chapter metadata. chapter_index is the zero-based chapter index.
per-program metadata. program_index is the zero-based program index.
If metadata specifier is omitted, it defaults to global.
By default, global metadata is copied from the first input file, per-stream and per-chapter metadata is copied along with streams/chapters. These default mappings are disabled by creating any mapping of the relevant type. A negative file index can be used to create a dummy mapping that just disables automatic copying.
For example to copy metadata from the first stream of the input file to global metadata of the output file:
ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3
To do the reverse, i.e. copy global metadata to all audio streams:
ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv
Note that simple 0 would work as well in this example, since global
metadata is assumed by default.
Copy chapters from input file with index input_file_index to the next output file. If no chapter mapping is specified, then chapters are copied from the first input file with at least one chapter. Use a negative file index to disable any chapter copying.
Show benchmarking information at the end of an encode. Shows CPU time used and maximum memory consumption. Maximum memory consumption is not supported on all systems, it will usually display as 0 if not supported.
Show benchmarking information during the encode. Shows CPU time used in various steps (audio/video encode/decode).
Exit after ffmpeg has been running for duration seconds.
Dump each input packet to stderr.
When dumping packets, also dump the payload.
Read input at native frame rate. Mainly used to simulate a grab device,
or live input stream (e.g. when reading from a file). Should not be used
with actual grab devices or live input streams (where it can cause packet
loss).
By default ffmpeg attempts to read the input(s) as fast as possible.
This option will slow down the reading of the input(s) to the native frame rate
of the input(s). It is useful for real-time output (e.g. live streaming).
Repeatedly loop output for formats that support looping such as animated GIF (0 will loop the output infinitely). This option is deprecated, use -loop.
Video sync method. For compatibility reasons old values can be specified as numbers. Newly added values will have to be specified as strings always.
Each frame is passed with its timestamp from the demuxer to the muxer.
Frames will be duplicated and dropped to achieve exactly the requested constant frame rate.
Frames are passed through with their timestamp or dropped so as to prevent 2 frames from having the same timestamp.
As passthrough but destroys all timestamps, making the muxer generate fresh timestamps based on frame-rate.
Chooses between 1 and 2 depending on muxer capabilities. This is the default method.
Note that the timestamps may be further modified by the muxer, after this. For example, in the case that the format option ‘avoid_negative_ts’ is enabled.
With -map you can select from which stream the timestamps should be taken. You can leave either video or audio unchanged and sync the remaining stream(s) to the unchanged one.
Frame drop threshold, which specifies how much behind video frames can be before they are dropped. In frame rate units, so 1.0 is one frame. The default is -1.1. One possible usecase is to avoid framedrops in case of noisy timestamps or to increase frame drop precision in case of exact timestamps.
Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps, the parameter is the maximum samples per second by which the audio is changed. -async 1 is a special case where only the start of the audio stream is corrected without any later correction.
Note that the timestamps may be further modified by the muxer, after this. For example, in the case that the format option ‘avoid_negative_ts’ is enabled.
This option has been deprecated. Use the aresample audio filter instead.
Do not process input timestamps, but keep their values without trying to sanitize them. In particular, do not remove the initial start time offset value.
Note that, depending on the ‘vsync’ option or on specific muxer processing (e.g. in case the format option ‘avoid_negative_ts’ is enabled) the output timestamps may mismatch with the input timestamps even when this option is selected.
When used with ‘copyts’, shift input timestamps so they start at zero.
This means that using e.g. -ss 50 will make output timestamps start at
50 seconds, regardless of what timestamp the input file started at.
Specify how to set the encoder timebase when stream copying. mode is an integer numeric value, and can assume one of the following values:
Use the demuxer timebase.
The time base is copied to the output encoder from the corresponding input demuxer. This is sometimes required to avoid non monotonically increasing timestamps when copying video streams with variable frame rate.
Use the decoder timebase.
The time base is copied to the output encoder from the corresponding input decoder.
Try to make the choice automatically, in order to generate a sane output.
Default value is -1.
Set the encoder timebase. timebase is a floating point number, and can assume one of the following values:
Assign a default value according to the media type.
For video - use 1/framerate, for audio - use 1/samplerate.
Use the input stream timebase when possible.
If an input stream is not available, the default timebase will be used.
Use the provided number as the timebase.
This field can be provided as a ratio of two integers (e.g. 1:24, 1:48000) or as a floating point number (e.g. 0.04166, 2.0833e-5)
Default value is 0.
Enable bitexact mode for (de)muxer and (de/en)coder
Finish encoding when the shortest input stream ends.
Timestamp discontinuity delta threshold.
Set the maximum demux-decode delay.
Set the initial demux-decode delay.
Assign a new stream-id value to an output stream. This option should be specified prior to the output filename to which it applies. For the situation where multiple output files exist, a streamid may be reassigned to a different value.
For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for an output mpegts file:
ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts
Set bitstream filters for matching streams. bitstream_filters is
a comma-separated list of bitstream filters. Use the -bsfs option
to get the list of bitstream filters.
ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
Force a tag/fourcc for matching streams.
Specify Timecode for writing. SEP is ’:’ for non drop timecode and ’;’ (or ’.’) for drop.
ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg
Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or outputs. For simple graphs – those with one input and one output of the same type – see the ‘-filter’ options. filtergraph is a description of the filtergraph, as described in the “Filtergraph syntax” section of the ffmpeg-filters manual.
Input link labels must refer to input streams using the
[file_index:stream_specifier] syntax (i.e. the same as ‘-map’
uses). If stream_specifier matches multiple streams, the first one will be
used. An unlabeled input will be connected to the first unused input stream of
the matching type.
Output link labels are referred to with ‘-map’. Unlabeled outputs are added to the first output file.
Note that with this option it is possible to use only lavfi sources without normal input files.
For example, to overlay an image over video
ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map '[out]' out.mkv
Here [0:v] refers to the first video stream in the first input file,
which is linked to the first (main) input of the overlay filter. Similarly the
first video stream in the second input is linked to the second (overlay) input
of overlay.
Assuming there is only one video stream in each input file, we can omit input labels, so the above is equivalent to
ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map '[out]' out.mkv
Furthermore we can omit the output label and the single output from the filter graph will be added to the output file automatically, so we can simply write
ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv
To generate 5 seconds of pure red video using lavfi color source:
ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv
Defines how many threads are used to process a filter_complex graph.
Similar to filter_threads but used for -filter_complex graphs only.
The default is the number of available CPUs.
Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or outputs. Equivalent to ‘-filter_complex’.
This option is similar to ‘-filter_complex’, the only difference is that its argument is the name of the file from which a complex filtergraph description is to be read.
This option enables or disables accurate seeking in input files with the ‘-ss’ option. It is enabled by default, so seeking is accurate when transcoding. Use ‘-noaccurate_seek’ to disable it, which may be useful e.g. when copying some streams and transcoding the others.
This option enables or disables seeking by timestamp in input files with the ‘-ss’ option. It is disabled by default. If enabled, the argument to the ‘-ss’ option is considered an actual timestamp, and is not offset by the start time of the file. This matters only for files which do not start from timestamp 0, such as transport streams.
This option sets the maximum number of queued packets when reading from the file or device. With low latency / high rate live streams, packets may be discarded if they are not read in a timely manner; raising this value can avoid it.
Print sdp information for an output stream to file. This allows dumping sdp information when at least one output isn’t an rtp stream. (Requires at least one of the output formats to be rtp).
Allows discarding specific streams or frames of streams at the demuxer. Not all demuxers support this.
Discard no frame.
Default, which discards no frames.
Discard all non-reference frames.
Discard all bidirectional frames.
Discard all frames excepts keyframes.
Discard all frames.
Stop and abort on various conditions. The following flags are available:
No packets were passed to the muxer, the output is empty.
Stop and exit on error
When transcoding audio and/or video streams, ffmpeg will not begin writing into the output until it has one packet for each such stream. While waiting for that to happen, packets for other streams are buffered. This option sets the size of this buffer, in packets, for the matching output stream.
The default value of this option should be high enough for most uses, so only touch this option if you are sure that you need it.
As a special exception, you can use a bitmap subtitle stream as input: it will be converted into a video with the same size as the largest video in the file, or 720x576 if no video is present. Note that this is an experimental and temporary solution. It will be removed once libavfilter has proper support for subtitles.
For example, to hardcode subtitles on top of a DVB-T recording stored in MPEG-TS format, delaying the subtitles by 1 second:
ffmpeg -i input.ts -filter_complex \ '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \ -sn -map '#0x2dc' output.mkv
(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video, audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)
A preset file contains a sequence of option=value pairs, one for each line, specifying a sequence of options which would be awkward to specify on the command line. Lines starting with the hash (’#’) character are ignored and are used to provide comments. Check the ‘presets’ directory in the FFmpeg source tree for examples.
There are two types of preset files: ffpreset and avpreset files.
ffpreset files are specified with the vpre, apre,
spre, and fpre options. The fpre option takes the
filename of the preset instead of a preset name as input and can be
used for any kind of codec. For the vpre, apre, and
spre options, the options specified in a preset file are
applied to the currently selected codec of the same type as the preset
option.
The argument passed to the vpre, apre, and spre
preset options identifies the preset file to use according to the
following rules:
First ffmpeg searches for a file named arg.ffpreset in the
directories ‘$FFMPEG_DATADIR’ (if set), and ‘$HOME/.ffmpeg’, and in
the datadir defined at configuration time (usually ‘PREFIX/share/ffmpeg’)
or in a ‘ffpresets’ folder along the executable on win32,
in that order. For example, if the argument is libvpx-1080p, it will
search for the file ‘libvpx-1080p.ffpreset’.
If no such file is found, then ffmpeg will search for a file named
codec_name-arg.ffpreset in the above-mentioned
directories, where codec_name is the name of the codec to which
the preset file options will be applied. For example, if you select
the video codec with -vcodec libvpx and use -vpre 1080p,
then it will search for the file ‘libvpx-1080p.ffpreset’.
avpreset files are specified with the pre option. They work similar to
ffpreset files, but they only allow encoder- specific options. Therefore, an
option=value pair specifying an encoder cannot be used.
When the pre option is specified, ffmpeg will look for files with the
suffix .avpreset in the directories ‘$AVCONV_DATADIR’ (if set), and
‘$HOME/.avconv’, and in the datadir defined at configuration time (usually
‘PREFIX/share/ffmpeg’), in that order.
First ffmpeg searches for a file named codec_name-arg.avpreset in
the above-mentioned directories, where codec_name is the name of the codec
to which the preset file options will be applied. For example, if you select the
video codec with -vcodec libvpx and use -pre 1080p, then it will
search for the file ‘libvpx-1080p.avpreset’.
If no such file is found, then ffmpeg will search for a file named arg.avpreset in the same directories.
If you specify the input format and device then ffmpeg can grab video and audio directly.
ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
Or with an ALSA audio source (mono input, card id 1) instead of OSS:
ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg
Note that you must activate the right video source and channel before launching ffmpeg with any TV viewer such as xawtv by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer.
Grab the X11 display with ffmpeg via
ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg
0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable.
ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg
0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. 10 is the x-offset and 20 the y-offset for the grabbing.
Any supported file format and protocol can serve as input to ffmpeg:
Examples:
ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
It will use the files:
/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V, /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the ‘-s’ option if ffmpeg cannot guess it.
ffmpeg -i /tmp/test.yuv /tmp/out.avi
test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed by the U and V planes at half vertical and horizontal resolution.
ffmpeg -i mydivx.avi hugefile.yuv
ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg.
ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
Converts a.wav to MPEG audio at 22050 Hz sample rate.
ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2
Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. ’-map file:index’ specifies which input stream is used for each output stream, in the order of the definition of output streams.
ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi
This is a typical DVD ripping example; the input is a VOB file, the
output an AVI file with MPEG-4 video and MP3 audio. Note that in this
command we use B-frames so the MPEG-4 stream is DivX5 compatible, and
GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
input video. Furthermore, the audio stream is MP3-encoded so you need
to enable LAME support by passing --enable-libmp3lame to configure.
The mapping is particularly useful for DVD transcoding
to get the desired audio language.
NOTE: To see the supported input formats, use ffmpeg -demuxers.
For extracting images from a video:
ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
This will extract one video frame per second from the video and will output them in files named ‘foo-001.jpeg’, ‘foo-002.jpeg’, etc. Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use the
above command in combination with the -frames:v or -t option,
or in combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
The syntax foo-%03d.jpeg specifies to use a decimal number
composed of three digits padded with zeroes to express the sequence
number. It is the same syntax supported by the C printf function, but
only formats accepting a normal integer are suitable.
When importing an image sequence, -i also supports expanding
shell-like wildcard patterns (globbing) internally, by selecting the
image2-specific -pattern_type glob option.
For example, for creating a video from filenames matching the glob pattern
foo-*.jpeg:
ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
The resulting output file ‘test12.nut’ will contain the first four streams from the input files in reverse order.
ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
This section documents the syntax and formats employed by the FFmpeg libraries and tools.
FFmpeg adopts the following quoting and escaping mechanism, unless explicitly specified. The following rules are applied:
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.
Crime d'Amour containing the ' special
character:
Crime d\'Amour
' needs to be escaped
when quoting it:
'Crime d'\''Amour'
' this string starts and ends with whitespaces '
' The string '\'string\'' is a string '
'c:\foo' can be written as c:\\foo
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.
There are two accepted syntaxes for expressing time duration.
[-][HH:]MM:SS[.m...]
HH expresses the number of hours, MM the number of minutes for a maximum of 2 digits, and SS the number of seconds for a maximum of 2 digits. The m at the end expresses decimal value for SS.
or
[-]S+[.m...]
S expresses the number of seconds, with the optional decimal part m.
In both expressions, the optional ‘-’ indicates negative duration.
The following examples are all valid time duration:
55 seconds
12 hours, 03 minutes and 45 seconds
23.189 seconds
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:
720x480
720x576
352x240
352x288
640x480
768x576
352x240
352x240
128x96
176x144
352x288
704x576
1408x1152
160x120
320x240
640x480
800x600
1024x768
1600x1200
2048x1536
1280x1024
2560x2048
5120x4096
852x480
1366x768
1600x1024
1920x1200
2560x1600
3200x2048
3840x2400
6400x4096
7680x4800
320x200
640x350
852x480
1280x720
1920x1080
2048x1080
1998x1080
2048x858
4096x2160
3996x2160
4096x1716
640x360
240x160
400x240
432x240
480x320
960x540
2048x1080
4096x2160
3840x2160
7680x4320
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:
30000/1001
25/1
30000/1001
25/1
30000/1001
25/1
24/1
24000/1001
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.
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:
0xF0F8FF
0xFAEBD7
0x00FFFF
0x7FFFD4
0xF0FFFF
0xF5F5DC
0xFFE4C4
0x000000
0xFFEBCD
0x0000FF
0x8A2BE2
0xA52A2A
0xDEB887
0x5F9EA0
0x7FFF00
0xD2691E
0xFF7F50
0x6495ED
0xFFF8DC
0xDC143C
0x00FFFF
0x00008B
0x008B8B
0xB8860B
0xA9A9A9
0x006400
0xBDB76B
0x8B008B
0x556B2F
0xFF8C00
0x9932CC
0x8B0000
0xE9967A
0x8FBC8F
0x483D8B
0x2F4F4F
0x00CED1
0x9400D3
0xFF1493
0x00BFFF
0x696969
0x1E90FF
0xB22222
0xFFFAF0
0x228B22
0xFF00FF
0xDCDCDC
0xF8F8FF
0xFFD700
0xDAA520
0x808080
0x008000
0xADFF2F
0xF0FFF0
0xFF69B4
0xCD5C5C
0x4B0082
0xFFFFF0
0xF0E68C
0xE6E6FA
0xFFF0F5
0x7CFC00
0xFFFACD
0xADD8E6
0xF08080
0xE0FFFF
0xFAFAD2
0x90EE90
0xD3D3D3
0xFFB6C1
0xFFA07A
0x20B2AA
0x87CEFA
0x778899
0xB0C4DE
0xFFFFE0
0x00FF00
0x32CD32
0xFAF0E6
0xFF00FF
0x800000
0x66CDAA
0x0000CD
0xBA55D3
0x9370D8
0x3CB371
0x7B68EE
0x00FA9A
0x48D1CC
0xC71585
0x191970
0xF5FFFA
0xFFE4E1
0xFFE4B5
0xFFDEAD
0x000080
0xFDF5E6
0x808000
0x6B8E23
0xFFA500
0xFF4500
0xDA70D6
0xEEE8AA
0x98FB98
0xAFEEEE
0xD87093
0xFFEFD5
0xFFDAB9
0xCD853F
0xFFC0CB
0xDDA0DD
0xB0E0E6
0x800080
0xFF0000
0xBC8F8F
0x4169E1
0x8B4513
0xFA8072
0xF4A460
0x2E8B57
0xFFF5EE
0xA0522D
0xC0C0C0
0x87CEEB
0x6A5ACD
0x708090
0xFFFAFA
0x00FF7F
0x4682B4
0xD2B48C
0x008080
0xD8BFD8
0xFF6347
0x40E0D0
0xEE82EE
0xF5DEB3
0xFFFFFF
0xF5F5F5
0xFFFF00
0x9ACD32
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:
front left
front right
front center
low frequency
back left
back right
front left-of-center
front right-of-center
back center
side left
side right
top center
top front left
top front center
top front right
top back left
top back center
top back right
downmix left
downmix right
wide left
wide right
surround direct left
surround direct right
low frequency 2
Standard channel layout compositions can be specified by using the following identifiers:
FC
FL+FR
FL+FR+LFE
FL+FR+FC
FL+FR+BC
FL+FR+FC+BC
FL+FR+BL+BR
FL+FR+SL+SR
FL+FR+FC+LFE
FL+FR+FC+BL+BR
FL+FR+FC+SL+SR
FL+FR+FC+LFE+BC
FL+FR+FC+LFE+BL+BR
FL+FR+FC+LFE+SL+SR
FL+FR+FC+BC+SL+SR
FL+FR+FLC+FRC+SL+SR
FL+FR+FC+BL+BR+BC
FL+FR+FC+LFE+BC+SL+SR
FL+FR+FC+LFE+BL+BR+BC
FL+FR+LFE+FLC+FRC+SL+SR
FL+FR+FC+BL+BR+SL+SR
FL+FR+FC+FLC+FRC+SL+SR
FL+FR+FC+LFE+BL+BR+SL+SR
FL+FR+FC+LFE+BL+BR+FLC+FRC
FL+FR+FC+LFE+FLC+FRC+SL+SR
FL+FR+FC+BL+BR+BC+SL+SR
DL+DR
A custom channel layout can be specified as a sequence of terms, separated by ’+’ or ’|’. Each term can be:
av_get_default_channel_layout). Note that not all channel counts have a
default layout.
AV_CH_* macros in ‘libavutil/channel_layout.h’.
Before libavutil version 53 the trailing character "c" to specify a number of channels was optional, but now it is required, while a channel layout mask can also be specified as a decimal number (if and only if not followed by "c" or "C").
See also the function av_get_channel_layout defined in
‘libavutil/channel_layout.h’.
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:
Compute absolute value of x.
Compute arccosine of x.
Compute arcsine of x.
Compute arctangent of x.
Compute principal value of the arc tangent of y/x.
Return 1 if x is greater than or equal to min and lesser than or equal to max, 0 otherwise.
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).
Round the value of expression expr upwards to the nearest integer. For example, "ceil(1.5)" is "2.0".
Return the value of x clipped between min and max.
Compute cosine of x.
Compute hyperbolic cosine of x.
Return 1 if x and y are equivalent, 0 otherwise.
Compute exponential of x (with base e, the Euler’s number).
Round the value of expression expr downwards to the nearest integer. For example, "floor(-1.5)" is "-2.0".
Compute Gauss function of x, corresponding to
exp(-x*x/2) / sqrt(2*PI).
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.
Return 1 if x is greater than y, 0 otherwise.
Return 1 if x is greater than or equal to y, 0 otherwise.
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.
Evaluate x, and if the result is non-zero return the result of the evaluation of y, return 0 otherwise.
Evaluate x, and if the result is non-zero return the evaluation result of y, otherwise the evaluation result of z.
Evaluate x, and if the result is zero return the result of the evaluation of y, return 0 otherwise.
Evaluate x, and if the result is zero return the evaluation result of y, otherwise the evaluation result of z.
Return 1.0 if x is +/-INFINITY, 0.0 otherwise.
Return 1.0 if x is NAN, 0.0 otherwise.
Load the value of the internal variable with number var, which was previously stored with st(var, expr). The function returns the loaded value.
Return linear interpolation between x and y by amount of z.
Compute natural logarithm of x.
Return 1 if x is lesser than y, 0 otherwise.
Return 1 if x is lesser than or equal to y, 0 otherwise.
Return the maximum between x and y.
Return the minimum between x and y.
Compute the remainder of division of x by y.
Return 1.0 if expr is zero, 0.0 otherwise.
Compute the power of x elevated y, it is equivalent to "(x)^(y)".
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
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.
Find an input value for which the function represented by expr with argument ld(0) is 0 in the interval 0..max.
The expression in expr must denote a continuous function or the result is undefined.
ld(0) is used to represent the function input value, which means
that the given expression will be evaluated multiple times with
various input values that the expression can access through
ld(0). When the expression evaluates to 0 then the
corresponding input value will be returned.
Round the value of expression expr to the nearest integer. For example, "round(1.5)" is "2.0".
Compute sine of x.
Compute hyperbolic sine of x.
Compute the square root of expr. This is equivalent to "(expr)^.5".
Compute expression 1/(1 + exp(4*x)).
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.
Compute tangent of x.
Compute hyperbolic tangent of x.
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.
Return the current (wallclock) time in seconds.
Round the value of expression expr towards zero to the nearest integer. For example, "trunc(-1.5)" is "-1.0".
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:
area of the unit disc, approximately 3.14
exp(1) (Euler’s number), approximately 2.718
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.
10^-24 / 2^-80
10^-21 / 2^-70
10^-18 / 2^-60
10^-15 / 2^-50
10^-12 / 2^-40
10^-9 / 2^-30
10^-6 / 2^-20
10^-3 / 2^-10
10^-2
10^-1
10^2
10^3 / 2^10
10^3 / 2^10
10^6 / 2^20
10^9 / 2^30
10^12 / 2^40
10^15 / 2^40
10^18 / 2^50
10^21 / 2^60
10^24 / 2^70
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:
Set bitrate in bits/s. Default value is 200K.
Set audio bitrate (in bits/s). Default value is 128K.
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.
Set generic flags.
Possible values:
Use four motion vector by macroblock (mpeg4).
Use 1/4 pel motion compensation.
Use loop filter.
Use fixed qscale.
Use internal 2pass ratecontrol in first pass mode.
Use internal 2pass ratecontrol in second pass mode.
Only decode/encode grayscale.
Do not draw edges.
Set error[?] variables during encoding.
Use interlaced DCT.
Force low delay.
Place global headers in extradata instead of every keyframe.
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.
Apply H263 advanced intra coding / mpeg4 ac prediction.
Deprecated, use mpegvideo private options instead.
Deprecated, use mpegvideo private options instead.
Apply interlaced motion estimation.
Use closed gop.
Set motion estimation method.
Possible values:
zero motion estimation (fastest)
full motion estimation (slowest)
EPZS motion estimation (default)
esa motion estimation (alias for full)
tesa motion estimation
dia motion estimation (alias for epzs)
log motion estimation
phods motion estimation
X1 motion estimation
hex motion estimation
umh motion estimation
iter motion estimation
Set extradata size.
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.
Set the group of picture (GOP) size. Default value is 12.
Set audio sampling rate (in Hz).
Set number of audio channels.
Set cutoff bandwidth. (Supported only by selected encoders, see their respective documentation sections.)
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.
Set the frame number.
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.
Set video quantizer scale blur (VBR).
Set min video quantizer scale (VBR). Must be included between -1 and 69, default value is 2.
Set max video quantizer scale (VBR). Must be included between -1 and 1024, default value is 31.
Set max difference between the quantizer scale (VBR).
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.
Set qp factor between P and B frames.
Set ratecontrol method.
Set strategy to choose between I/P/B-frames.
Set RTP payload size in bytes.
Workaround not auto detected encoder bugs.
Possible values:
some old lavc generated msmpeg4v3 files (no autodetection)
Xvid interlacing bug (autodetected if fourcc==XVIX)
(autodetected if fourcc==UMP4)
padding bug (autodetected)
illegal vlc bug (autodetected per fourcc)
old standard qpel (autodetected per fourcc/version)
direct-qpel-blocksize bug (autodetected per fourcc/version)
edge padding bug (autodetected per fourcc/version)
Workaround various bugs in microsoft broken decoders.
trancated frames
Set single coefficient elimination threshold for luminance (negative values also consider DC coefficient).
Set single coefficient elimination threshold for chrominance (negative values also consider dc coefficient)
Specify how strictly to follow the standards.
Possible values:
strictly conform to an older more strict version of the spec or reference software
strictly conform to all the things in the spec no matter what consequences
allow unofficial extensions
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.
Set QP offset between P and B frames.
Set error detection flags.
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
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.
consider things that violate the spec and have not been seen in the wild as errors
consider all spec non compliancies as errors
consider things that a sane encoder should not do as an error
Use MPEG quantizers instead of H.263.
How to keep quantizer between qmin and qmax (0 = clip, 1 = use differentiable function).
Set experimental quantizer modulation.
Set experimental quantizer modulation.
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.
Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use elsewise.
Set ratecontrol buffer size (in bits).
Currently useless.
Set QP factor between P and I frames.
Set QP offset between P and I frames.
Set initial complexity for 1-pass encoding.
Set DCT algorithm.
Possible values:
autoselect a good one (default)
fast integer
accurate integer
floating point AAN DCT
Compress bright areas stronger than medium ones.
Set temporal complexity masking.
Set spatial complexity masking.
Set inter masking.
Compress dark areas stronger than medium ones.
Select IDCT implementation.
Possible values:
Automatically pick a IDCT compatible with the simple one
floating point AAN IDCT
Set error concealment strategy.
Possible values:
iterative motion vector (MV) search (slow)
use strong deblock filter for damaged MBs
favor predicting from the previous frame instead of the current
Set prediction method.
Possible values:
Set sample aspect ratio.
Set sample aspect ratio. Alias to aspect.
Print specific debug info.
Possible values:
picture info
rate control
macroblock (MB) type
per-block quantization parameter (QP)
display complexity metadata for the upcoming frame, GoP or for a given duration.
error recognition
memory management control operations (H.264)
picture buffer allocations
threading operations
skip motion compensation
Set full pel me compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set sub pel me compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set macroblock compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set interlaced dct compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set diamond type & size for motion estimation.
Set amount of motion predictors from the previous frame.
Set pre motion estimation.
Set pre motion estimation compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set diamond type & size for motion estimation pre-pass.
Set sub pel motion estimation quality.
Set limit motion vectors range (1023 for DivX player).
Set intra quant bias.
Set inter quant bias.
Possible values:
variable length coder / huffman coder
arithmetic coder
raw (no encoding)
run-length coder
deflate-based coder
Set context model.
Set macroblock decision algorithm (high quality mode).
Possible values:
use mbcmp (default)
use fewest bits
use best rate distortion
Set scene change threshold.
Set min lagrange factor (VBR).
Set max lagrange factor (VBR).
Set noise reduction.
Set number of bits which should be loaded into the rc buffer before decoding starts.
Possible values:
Allow non spec compliant speedup tricks.
Deprecated, use mpegvideo private options instead.
Skip bitstream encoding.
Ignore cropping information from sps.
Place global headers at every keyframe instead of in extradata.
Frame data might be split into multiple chunks.
Show all frames before the first keyframe.
Deprecated, use mpegvideo private options instead.
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’.
Deprecated, use mpegvideo private options instead.
Set the number of threads to be used, in case the selected codec implementation supports multi-threading.
Possible values:
automatically select the number of threads to set
Default value is ‘auto’.
Set motion estimation threshold.
Set macroblock threshold.
Set intra_dc_precision.
Set nsse weight.
Set number of macroblock rows at the top which are skipped.
Set number of macroblock rows at the bottom which are skipped.
Possible values:
Possible values:
Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
Set frame skip threshold.
Set frame skip factor.
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.
Set frame skip compare function.
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Increase the quantizer for macroblocks close to borders.
Set min macroblock lagrange factor (VBR).
Set max macroblock lagrange factor (VBR).
Set motion estimation bitrate penalty compensation (1.0 = 256).
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:
Discard no frame.
Discard useless frames like 0-sized frames.
Discard all non-reference frames.
Discard all bidirectional frames.
Discard all frames excepts keyframes.
Discard all frames.
Default value is ‘default’.
Refine the two motion vectors used in bidirectional macroblocks.
Downscale frames for dynamic B-frame decision.
Set minimum interval between IDR-frames.
Set reference frames to consider for motion compensation.
Set chroma qp offset from luma.
Set rate-distortion optimal quantization.
Set value multiplied by qscale for each frame and added to scene_change_score.
Adjust sensitivity of b_frame_strategy 1.
Set GOP timecode frame start number, in non drop frame format.
Set desired number of audio channels.
Possible values:
Possible values:
Possible values:
BT.709
BT.470 M
BT.470 BG
SMPTE 170 M
SMPTE 240 M
Film
BT.2020
SMPTE ST 428-1
SMPTE 431-2
SMPTE 432-1
JEDEC P22
Possible values:
BT.709
BT.470 M
BT.470 BG
SMPTE 170 M
SMPTE 240 M
Linear
Log
Log square root
IEC 61966-2-4
BT.1361
IEC 61966-2-1
BT.2020 - 10 bit
BT.2020 - 12 bit
SMPTE ST 2084
SMPTE ST 428-1
ARIB STD-B67
Possible values:
RGB
BT.709
FCC
BT.470 BG
SMPTE 170 M
SMPTE 240 M
YCOCG
BT.2020 NCL
BT.2020 CL
SMPTE 2085
If used as input parameter, it serves as a hint to the decoder, which color_range the input has. Possible values:
MPEG (219*2^(n-8))
JPEG (2^n-1)
Possible values:
Set the log level offset.
Number of slices, used in parallelized encoding.
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:
Decode more than one part of a single frame at once.
Multithreading using slices works only when the video was encoded with slices.
Decode more than one frame at once.
Default value is ‘slice+frame’.
Set audio service type.
Possible values:
Main Audio Service
Effects
Visually Impaired
Hearing Impaired
Dialogue
Commentary
Emergency
Voice Over
Karaoke
Set sample format audio decoders should prefer. Default value is
none.
Set the input subtitles character encoding.
Set/override the field order of the video. Possible values:
Progressive video
Interlaced video, top field coded and displayed first
Interlaced video, bottom field coded and displayed first
Interlaced video, top coded first, bottom displayed first
Interlaced video, bottom coded first, top displayed first
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.
"," separated list of allowed decoders. By default all are allowed.
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
Maximum number of pixels per image. This value can be used to avoid out of memory failures due to large images.
Enable cropping if cropping parameters are multiples of the required
alignment for the left and top parameters. If the alignment is not met the
cropping will be partially applied to maintain alignment.
Default is 1 (enabled).
Note: The required alignment depends on if AV_CODEC_FLAG_UNALIGNED is set and the
CPU. AV_CODEC_FLAG_UNALIGNED cannot be changed from the command line. Also hardware
decoders will not apply left/top Cropping.
Decoders 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.
A description of some of the currently available video decoders follows.
Raw video decoder.
This decoder decodes rawvideo streams.
Specify the assumed field type of the input video.
the video is assumed to be progressive (default)
bottom-field-first is assumed
top-field-first is assumed
A description of some of the currently available audio decoders follows.
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).
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 disabled. Produces full range audio.
DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full range and full compression.
DRC enabled. Applies drc_scale asymmetrically. Loud sounds are fully compressed. Soft sounds are enhanced.
FLAC audio decoder.
This decoder aims to implement the complete FLAC specification from Xiph.
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.
Internal wave synthesizer.
This decoder generates wave patterns according to predefined sequences. Its use is purely internal and the format of the data it accepts is not publicly documented.
libcelt 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 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 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.
The following option is supported by the libilbc wrapper.
Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled).
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 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 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.
Compute clut if no matching CLUT is in the stream.
Never compute CLUT
Always compute CLUT and override the one provided in the stream.
Selects the dvb substream, or all substreams if -1 which is default.
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.
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.
Specify the IFO file from which the global palette is obtained. (experimental)
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 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.
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 *.
Discards the top teletext line. Default value is 1.
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.
X offset of generated bitmaps, default is 0.
Y offset of generated bitmaps, default is 0.
Chops leading and trailing spaces and removes empty lines from the generated text. This option is useful for teletext based subtitles where empty spaces may be present at the start or at the end of the lines or empty lines may be present between the subtitle lines because of double-sized teletext characters. Default value is 1.
Sets the display duration of the decoded teletext pages or subtitles in milliseconds. Default value is 30000 which is 30 seconds.
Force transparent background of the generated teletext bitmaps. Default value is 0 which means an opaque background.
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 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.
A description of some of the currently available audio encoders follows.
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.
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.
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’.
Set cutoff frequency. If unspecified will allow the encoder to dynamically adjust the cutoff to improve clarity on low bitrates.
Set AAC encoder coding method. Possible values:
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.
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.
Constant quantizer method.
Uses a cheaper version of twoloop algorithm that doesn’t try to do as many clever adjustments. Worse with low bitrates (less than 64kbps), but is better and much faster at higher bitrates. This is the default choice for a coder
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".
Sets intensity stereo coding tool usage. By default, it’s enabled and will automatically toggle IS for similar pairs of stereo bands if it’s beneficial. Can be disabled for debugging by setting the value to "disable".
Uses perceptual noise substitution to replace low entropy high frequency bands with imperceptible white noise during the decoding process. By default, it’s enabled, but can be disabled for debugging purposes by using "disable".
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".
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.
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.
Sets the encoding profile, possible values:
The default, AAC "Low-complexity" profile. Is the most compatible and produces decent quality.
Equivalent to -profile:a aac_low -aac_pns 0. PNS was introduced with the
MPEG4 specifications.
Long term prediction profile, is enabled by and will enable the ‘aac_ltp’ option. Introduced in MPEG4.
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’.
AC-3 audio encoders.
These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet).
The ac3 encoder uses floating-point math, while the ac3_fixed
encoder only uses fixed-point integer math. This does not mean that one is
always faster, just that one or the other may be better suited to a
particular system. The floating-point encoder will generally produce better
quality audio for a given bitrate. The ac3_fixed encoder is not the
default codec for any of the output formats, so it must be specified explicitly
using the option -acodec ac3_fixed in order to use it.
The AC-3 metadata options are used to set parameters that describe the audio, but in most cases do not affect the audio encoding itself. Some of the options do directly affect or influence the decoding and playback of the resulting bitstream, while others are just for informational purposes. A few of the options will add bits to the output stream that could otherwise be used for audio data, and will thus affect the quality of the output. Those will be indicated accordingly with a note in the option list below.
These parameters are described in detail in several publicly-available documents.
Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.
The metadata values set at initialization will be used for every frame in the stream. (default)
Metadata values can be changed before encoding each frame.
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:
Apply -3dB gain
Apply -4.5dB gain (default)
Apply -6dB gain
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:
Apply -3dB gain
Apply -6dB gain (default)
Silence Surround Channel(s)
Audio Production Information is optional information describing the mixing environment. Either none or both of the fields are written to the bitstream.
Mixing Level. 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. 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.
Not Indicated (default)
Large Room
Small Room
Copyright Indicator. Specifies whether a copyright exists for this audio.
No Copyright Exists (default)
Copyright Exists
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.
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.
Not Indicated (default)
Not Dolby Surround Encoded
Dolby Surround Encoded
Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy.
Not Original Source
Original Source (default)
The extended bitstream options are part of the Alternate Bit Stream Syntax as
specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
If any one parameter in a group is specified, all values in that group will be
written to the bitstream. Default values are used for those that are written
but have not been specified. If the mixing levels are written, the decoder
will use these values instead of the ones specified in the center_mixlev
and surround_mixlev options if it supports the Alternate Bit Stream
Syntax.
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.
Not Indicated (default)
Lt/Rt Downmix Preferred
Lo/Ro Downmix Preferred
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.
Apply +3dB gain
Apply +1.5dB gain
Apply 0dB gain
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain (default)
Apply -6.0dB gain
Silence Center Channel
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.
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain
Apply -6.0dB gain (default)
Silence Surround Channel(s)
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.
Apply +3dB gain
Apply +1.5dB gain
Apply 0dB gain
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain (default)
Apply -6.0dB gain
Silence Center Channel
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.
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain
Apply -6.0dB gain (default)
Silence Surround Channel(s)
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.
Not Indicated (default)
Dolby Surround EX Off
Dolby Surround EX On
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.
Not Indicated (default)
Dolby Headphone Off
Dolby Headphone On
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.
Standard A/D Converter (default)
HDCD A/D Converter
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.
Set lowpass cutoff frequency. If unspecified, the encoder selects a default determined by various other encoding parameters.
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.
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.
Selected by Encoder (default)
Disable Channel Coupling
Enable Channel Coupling
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.
Selected by Encoder (default)
FLAC (Free Lossless Audio Codec) Encoder
The following options are supported by FFmpeg’s flac encoder.
Sets the compression level, which chooses defaults for many other options if they are not set explicitly. Valid values are from 0 to 12, 5 is the default.
Sets the size of the frames in samples per channel.
Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the default.
Sets the first stage LPC algorithm
LPC is not used
fixed LPC coefficients
Number of passes to use for Cholesky factorization during LPC analysis
The minimum partition order
The maximum partition order
Bruteforce search
Channel mode
The mode is chosen automatically for each frame
Channels are independently coded
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 Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is applied after the first stage to finetune the coefficients. This is quite slow and slightly improves compression.
Opus encoder.
This is a native FFmpeg encoder for the Opus format. Currently its in development and only implements the CELT part of the codec. Its quality is usually worse and at best is equal to the libopus encoder.
Set bit rate in bits/s. If unspecified it uses the number of channels and the layout to make a good guess.
Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly decrease quality.
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/.
The following options are mapped on the shared FFmpeg codec options.
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.
Set audio sampling rate (in Hz).
Set the number of audio channels.
Enable fixed quality, VBR (Variable Bit Rate) mode. Note that VBR is implicitly enabled when the ‘vbr’ value is positive.
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.
Set audio profile.
The following profiles are recognized:
Low Complexity AAC (LC)
High Efficiency AAC (HE-AAC)
High Efficiency AAC version 2 (HE-AACv2)
Low Delay AAC (LD)
Enhanced Low Delay AAC (ELD)
If not specified it is set to ‘aac_low’.
The following are private options of the libfdk_aac encoder.
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.
Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled if set to 0.
Default value is 0.
Set SBR/PS signaling style.
It can assume one of the following values:
choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)
implicit backwards compatible signaling
explicit SBR, implicit PS signaling
explicit hierarchical signaling
Default value is ‘default’.
Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.
Default value is 0.
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.
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:
32 kbps/channel
40 kbps/channel
48-56 kbps/channel
64 kbps/channel
about 80-96 kbps/channel
Default value is 0.
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
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
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.
The following options are supported by the libmp3lame wrapper. The
lame-equivalent of the options are listed in parentheses.
Set bitrate expressed in bits/s for CBR or ABR. LAME bitrate is
expressed in kilobits/s.
Set constant quality setting for VBR. This option is valid only
using the ffmpeg command-line tool. For library interface
users, use ‘global_quality’.
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.
Set lowpass cutoff frequency. If unspecified, the encoder dynamically adjusts the cutoff.
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.
Enable the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo. Default value is 1.
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.
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.
Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).
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.
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.
Set the bit rate in bits/s. FFmpeg’s ‘b’ option is
expressed in bits/s, while opusenc’s ‘bitrate’ in
kilobits/s.
Set VBR mode. The FFmpeg ‘vbr’ option has the following
valid arguments, with the opusenc equivalent options
in parentheses:
Use constant bit rate encoding.
Use variable bit rate encoding (the default).
Use constrained variable bit rate encoding.
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.
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.
Set expected packet loss percentage. The default is 0.
Set intended application type. Valid options are listed below:
Favor improved speech intelligibility.
Favor faithfulness to the input (the default).
Restrict to only the lowest delay modes.
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).
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.
If set to 0, disables the use of phase inversion for intensity stereo, improving the quality of mono downmixes, but slightly reducing normal stereo quality. The default is 1 (phase inversion enabled).
Shine Fixed-Point MP3 encoder wrapper.
Shine is a fixed-point MP3 encoder. It has a far better performance on platforms without an FPU, e.g. armel CPUs, and some phones and tablets. However, as it is more targeted on performance than quality, it is not on par with LAME and other production-grade encoders quality-wise. Also, according to the project’s homepage, this encoder may not be free of bugs as the code was written a long time ago and the project was dead for at least 5 years.
This encoder only supports stereo and mono input. This is also CBR-only.
The original project (last updated in early 2007) is at http://sourceforge.net/projects/libshine-fxp/. We only support the updated fork by the Savonet/Liquidsoap project at https://github.com/savonet/shine.
Requires the presence of the libshine headers and library during
configuration. You need to explicitly configure the build with
--enable-libshine.
See also libmp3lame.
The following options are supported by the libshine wrapper. The
shineenc-equivalent of the options are listed in parentheses.
Set bitrate expressed in bits/s for CBR. shineenc ‘-b’ option
is expressed in kilobits/s.
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.
The following options are supported by the libtwolame wrapper. The
twolame-equivalent options follow the FFmpeg ones and are in
parentheses.
Set bitrate expressed in bits/s for CBR. twolame ‘b’
option is expressed in kilobits/s. Default value is 128k.
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’.
Set the mode of the resulting audio. Possible values:
Choose mode automatically based on the input. This is the default.
Stereo
Joint stereo
Dual channel
Mono
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.
Enable energy levels extensions when set to 1. The default value is 0 (disabled).
Enable CRC error protection when set to 1. The default value is 0 (disabled).
Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).
Set MPEG audio original flag when set to 1. The default value is 0 (disabled).
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.
Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).
libvorbis encoder wrapper.
Requires the presence of the libvorbisenc headers and library during
configuration. You need to explicitly configure the build with
--enable-libvorbis.
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).
Set bitrate expressed in bits/s for ABR. oggenc ‘-b’ is
expressed in kilobits/s.
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’.
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).
Set minimum bitrate expressed in bits/s. oggenc ‘-m’ is
expressed in kilobits/s.
Set maximum bitrate expressed in bits/s. oggenc ‘-M’ is
expressed in kilobits/s. This only has effect on ABR mode.
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.
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.
wavpack command line utility’s corresponding options are listed in
parentheses, if any.
Default is 32768.
Set speed vs. compression tradeoff. Acceptable arguments are listed below:
Fast mode.
Normal (default) settings.
High quality.
Very high quality.
Same as ‘3’, but with extra processing enabled.
‘4’ is the same as ‘-x2’ and ‘8’ is the same as ‘-x6’.
Motion JPEG encoder.
Set the huffman encoding strategy. Possible values:
Use the default huffman tables. This is the default strategy.
Compute and use optimal huffman tables.
WavPack lossless audio encoder.
This is a libavcodec-native WavPack encoder. There is also an encoder based on libwavpack, but there is virtually no reason to use that encoder.
See also libwavpack.
The equivalent options for wavpack command line utility are listed in
parentheses.
The following shared options are effective for this encoder. Only special notes about this particular encoder will be documented here. For the general meaning of the options, see the Codec Options chapter.
For this encoder, the range for this option is between 128 and 131072. Default is automatically decided based on sample rate and number of channel.
For the complete formula of calculating default, see ‘libavcodec/wavpackenc.c’.
This option’s syntax is consistent with libwavpack’s.
Set whether to enable joint stereo. Valid values are:
Force mid/side audio encoding.
Force left/right audio encoding.
Let the encoder decide automatically.
Set whether to enable optimization for mono. This option is only effective for non-mono streams. Available values:
enabled
disabled
A description of some of the currently available video encoders follows.
Vidvox Hap video encoder.
Specifies the Hap format to encode.
Default value is ‘hap’.
Specifies the number of chunks to split frames into, between 1 and 64. This permits multithreaded decoding of large frames, potentially at the cost of data-rate. The encoder may modify this value to divide frames evenly.
Default value is 1.
Specifies the second-stage compressor to use. If set to ‘none’, ‘chunks’ will be limited to 1, as chunked uncompressed frames offer no benefit.
Default value is ‘snappy’.
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.
Can be set to either j2k or jp2 (the default) that
makes it possible to store non-rgb pix_fmts.
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’.
Set target video bitrate in bit/s and enable rate control.
Set kvazaar parameters as a list of name=value pairs separated by commas (,). See kvazaar documentation for a list of options.
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.
The following FFmpeg global options affect the configurations of the libopenh264 encoder.
Set the bitrate (as a number of bits per second).
Set the GOP size.
Set the max bitrate (as a number of bits per second).
Set global header in the bitstream.
Set the number of slices, used in parallelized encoding. Default value is 0. This is only used when ‘slice_mode’ is set to ‘fixed’.
Set slice mode. Can assume one of the following possible values:
a fixed number of slices
one slice per row of macroblocks
automatic number of slices according to number of threads
dynamic slicing
Default value is ‘auto’.
Enable loop filter, if set to 1 (automatically enabled). To disable set a value of 0.
Set profile restrictions. If set to the value of ‘main’ enable
CABAC (set the SEncParamExt.iEntropyCodingModeFlag flag to 1).
Set maximum NAL size in bytes.
Allow skipping frames to hit the target bitrate if set to 1.
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/.
The following global options are mapped to internal libtheora options which affect the quality and the bitrate of the encoded stream.
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.
Used to enable constant quality mode (VBR) encoding through the
‘qscale’ flag, and to enable the pass1 and pass2
modes.
Set the GOP size.
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.
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’.
ffmpeg:
ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
ffmpeg to convert a CBR 1000 kbps Theora video stream:
ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
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.
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.
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.
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.
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.
Set datarate undershoot (min) percentage of the target bitrate.
Set datarate overshoot (max) percentage of the target bitrate.
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).
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 == bitrate).
Use best quality deadline. Poorly named and quite slow, this option should be avoided as it may give worse quality output than good.
Use good quality deadline. This is a good trade-off between speed and quality when used with the ‘cpu-used’ option.
Use realtime quality deadline.
Set quality/speed ratio modifier. Higher values speed up the encode at the cost of quality.
Set a change threshold on blocks below which they will be skipped by the encoder.
Note that FFmpeg’s ‘slices’ option gives the total number of partitions,
while vpxenc’s ‘token-parts’ is given as
log2(partitions).
Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0 means unlimited.
VPX_EFLAG_FORCE_KF
Enable use of alternate reference frames (2-pass only).
Set altref noise reduction max frame count.
Set altref noise reduction filter type: backward, forward, centered.
Set altref noise reduction filter strength.
Set number of frames to look ahead for frametype and ratecontrol.
Enable error resiliency features.
Enable lossless mode.
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.
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.
Enable frame parallel decodability features.
Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3: cyclic refresh, 4: equator360).
Set input color space. The VP9 bitstream supports signaling the following colorspaces:
Enable row based multi-threading.
Set content type: default (0), screen (1), film (2).
Corpus VBR mode is a variant of standard VBR where the complexity distribution midpoint is passed in rather than calculated for a specific clip or chunk.
The valid range is [0, 10000]. 0 (default) uses standard VBR.
For more information about libvpx see: http://www.webmproject.org/
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.
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.
Enables/Disables use of lossless mode. Default is 0.
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.
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.
Configuration preset. This does some automatic settings based on the general type of the image.
Do not use a preset.
Use the encoder default.
Digital picture, like portrait, inner shot
Outdoor photograph, with natural lighting
Hand or line drawing, with high-contrast details
Small-sized colorful images
Text-like
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.
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.
The following options are supported by the libx264 wrapper. The
x264-equivalent options or values are listed in parentheses
for easy migration.
To reduce the duplication of documentation, only the private options and some others requiring special attention are documented here. For the documentation of the undocumented generic options, see the Codec Options chapter.
To get a more accurate and extensive documentation of the libx264
options, invoke the command x264 --fullhelp or consult
the libx264 documentation.
Set bitrate in bits/s. Note that FFmpeg’s ‘b’ option is
expressed in bits/s, while x264’s ‘bitrate’ is in
kilobits/s.
Minimum quantizer scale.
Maximum quantizer scale.
Maximum difference between quantizer scales.
Quantizer curve blur
Quantizer curve compression factor
Number of reference frames each P-frame can use. The range is from 0-16.
Sets the threshold for the scene change detection.
Performs Trellis quantization to increase efficiency. Enabled by default.
Maximum range of the motion search in pixels.
Set motion estimation method. Possible values in the decreasing order of speed:
Diamond search with radius 1 (fastest). ‘epzs’ is an alias for ‘dia’.
Hexagonal search with radius 2.
Uneven multi-hexagon search.
Exhaustive search.
Hadamard exhaustive search (slowest).
Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces it to choose an IDR-frame.
Sub-pixel motion estimation method.
Adaptive B-frame placement decision algorithm. Use only on first-pass.
Minimum GOP size.
Set entropy encoder. Possible values:
Enable CABAC.
Enable CAVLC and disable CABAC. It generates the same effect as
x264’s ‘--no-cabac’ option.
Set full pixel motion estimation comparison algorithm. Possible values:
Enable chroma in motion estimation.
Ignore chroma in motion estimation. It generates the same effect as
x264’s ‘--no-chroma-me’ option.
Number of encoding threads.
Set multithreading technique. Possible values:
Slice-based multithreading. It generates the same effect as
x264’s ‘--sliced-threads’ option.
Frame-based multithreading.
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.
Set the encoding preset.
Set tuning of the encoding params.
Set profile restrictions.
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.
Set the quality for constant quality mode.
In CRF mode, prevents VBV from lowering quality beyond this point.
Set constant quantization rate control method parameter.
Set AQ method. Possible values:
Disabled.
Variance AQ (complexity mask).
Auto-variance AQ (experimental).
Set AQ strength, reduce blocking and blurring in flat and textured areas.
Use psychovisual optimizations when set to 1. When set to 0, it has the
same effect as x264’s ‘--no-psy’ option.
Set strength of psychovisual optimization, in psy-rd:psy-trellis format.
Set number of frames to look ahead for frametype and ratecontrol.
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.
Set weighted prediction method for P-frames. Possible values:
Disabled
Enable only weighted refs
Enable both weighted refs and duplicates
Enable calculation and printing SSIM stats after the encoding.
Enable the use of Periodic Intra Refresh instead of IDR frames when set to 1.
Configure the encoder to generate AVC-Intra. Valid values are 50,100 and 200
Configure the encoder to be compatible with the bluray standard. It is a shorthand for setting "bluray-compat=1 force-cfr=1".
Set the influence on how often B-frames are used.
Set method for keeping of some B-frames as references. Possible values:
Disabled.
Strictly hierarchical pyramid.
Non-strict (not Blu-ray compatible).
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.
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.
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.
Enable use of access unit delimiters when set to 1.
Enable use macroblock tree ratecontrol when set to 1. When set
to 0, it has the same effect as x264’s
‘--no-mbtree’ option.
Set loop filter parameters, in alpha:beta form.
Set fluctuations reduction in QP (before curve compression).
Set partitions to consider as a comma-separated list of. Possible values in the list:
8x8 P-frame partition.
4x4 P-frame partition.
4x4 B-frame partition.
8x8 I-frame partition.
4x4 I-frame partition. (Enabling ‘p4x4’ requires ‘p8x8’ to be enabled. Enabling ‘i8x8’ requires adaptive spatial transform (‘8x8dct’ option) to be enabled.)
Do not consider any partitions.
Consider every partition.
Set direct MV prediction mode. Possible values:
Disable MV prediction.
Enable spatial predicting.
Enable temporal predicting.
Automatically decided.
Set the limit of the size of each slice in bytes. If not specified but RTP payload size (‘ps’) is specified, that is used.
Set the file name for multi-pass stats.
Set signal HRD information (requires ‘vbv-bufsize’ to be set). Possible values:
Disable HRD information signaling.
Variable bit rate.
Constant bit rate (not allowed in MP4 container).
Set any x264 option, see x264 --fullhelp for a list.
Argument is a list of key=value couples separated by ":". In filter and psy-rd options that use ":" as a separator themselves, use "," instead. They accept it as well since long ago but this is kept undocumented for some reason.
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
Import closed captions (which must be ATSC compatible format) into output. Only the mpeg2 and h264 decoders provide these. Default is 1 (on).
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).
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’.
Set the x265 preset.
Set the x265 tune parameter.
Set profile restrictions.
Set the quality for constant quality mode.
Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces it to choose an IDR-frame.
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
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.
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.
Set specific encoding flags. Possible values:
Use four motion vector by macroblock.
Enable high quality AC prediction.
Only encode grayscale.
Enable the use of global motion compensation (GMC).
Enable quarter-pixel motion compensation.
Enable closed GOP.
Place global headers in extradata instead of every keyframe.
Set motion estimation method. Possible values in decreasing order of speed and increasing order of quality:
Use no motion estimation (default).
Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks. ‘x1’ and ‘log’ are aliases for ‘phods’.
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.
Enable all of the things described above, plus extended 16x16 and 8x8 blocks search.
Set macroblock decision algorithm. Possible values in the increasing order of quality:
Use macroblock comparing function algorithm (default).
Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks.
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.
Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled).
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.
Set structural similarity (SSIM) displaying method. Possible values:
Disable displaying of SSIM information.
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).
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).
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.
MPEG-2 video encoder.
Specifies if the encoder should write a sequence_display_extension to the output.
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.
Never write it.
Always write it.
Specifies the video_format written into the sequence display extension indicating the source of the video pictures. The default is ‘unspecified’, can be ‘component’, ‘pal’, ‘ntsc’, ‘secam’ or ‘mac’. For maximum compatibility, use ‘component’.
PNG image encoder.
Set physical density of pixels, in dots per inch, unset by default
Set physical density of pixels, in dots per meter, unset by default
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.
Select the ProRes profile to encode
Select quantization matrix.
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.
How many bits to allot for coding one macroblock. Different profiles use between 200 and 2400 bits per macroblock, the maximum is 8000.
Number of macroblocks in each slice (1-8); the default value (8) should be good in almost all situations.
Override the 4-byte vendor ID. A custom vendor ID like apl0 would claim the stream was produced by the Apple encoder.
Specify number of bits for alpha component. Possible values are 0, 8 and 16. Use 0 to disable alpha plane coding.
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.
The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)
The ratecontrol method is selected as follows:
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:
dia size for the iterative motion estimation
Wrappers for hardware encoders accessible via VAAPI.
These encoders only accept input in VAAPI hardware surfaces. If you have input in software frames, use the ‘hwupload’ filter to upload them to the GPU.
The following standard libavcodec options are used:
Speed / quality tradeoff: higher values are faster / worse quality.
Size / quality tradeoff: higher values are smaller / worse quality.
‘profile’ sets the value of profile_idc and the constraint_set*_flags. ‘level’ sets the value of level_idc.
Use low-power encoding mode.
Set entropy encoder (default is cabac). Possible values:
Use CABAC.
Use CAVLC.
‘profile’ and ‘level’ set the values of general_profile_idc and general_level_idc respectively.
Always encodes using the standard quantisation and huffman tables - ‘global_quality’ scales the standard quantisation table (range 1-100).
‘profile’ and ‘level’ set the value of profile_and_level_indication.
No rate control is supported.
B-frames are not supported.
‘global_quality’ sets the q_idx used for non-key frames (range 0-127).
Manually set the loop filter parameters.
‘global_quality’ sets the q_idx used for P-frames (range 0-255).
Manually set the loop filter parameters.
B-frames are supported, but the output stream is always in encode order rather than display order. If B-frames are enabled, it may be necessary to use the ‘vp9_raw_reorder’ bitstream filter to modify the output stream to display frames in the correct order.
Only normal frames are produced - the ‘vp9_superframe’ bitstream filter may be required to produce a stream usable with all decoders.
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).
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.
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.
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.
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.
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.
Sets the undershoot tolerance of the rate control system in percent. This is to prevent an expensive search from being run.
Sets the quantization matrix preset to use by default or when ‘wavelet_depth’ is set to 5
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.
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.
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.
Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration bitstream.
This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS header and removes the ADTS header.
This filter is required for example when copying an AAC stream from a raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or to MOV/MP4 files and related formats such as 3GP or M4A. Please note that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
Remove zero padding at the end of a packet.
Extract the core from a DCA/DTS stream, dropping extensions such as DTS-HD.
Add extradata to the beginning of the filtered packets.
The additional argument specifies which packets should be filtered. It accepts the values:
add extradata to all key packets
add extradata to all packets
If not specified it is assumed ‘e’.
For example the following ffmpeg command forces a global
header (thus disabling individual packet headers) in the H.264 packets
generated by the libx264 encoder, but corrects them by adding
the header stored in extradata to the key packets:
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
Extract the core from a E-AC-3 stream, dropping extra channels.
Extract the in-band extradata.
Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers, or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-band" (i.e. as a part of the bitstream containing the coded frames) or "out of band" (e.g. on the container level). This latter form is called "extradata" in FFmpeg terminology.
This bitstream filter detects the in-band headers and makes them available as extradata.
When this option is enabled, the long-term headers are removed from the bitstream after extraction.
Remove units with types in or not in a given set from the stream.
List of unit types or ranges of unit types to pass through while removing all others. This is specified as a ’|’-separated list of unit type values or ranges of values with ’-’.
Identical to ‘pass_types’, except the units in the given set removed and all others passed through.
Extradata is unchanged by this transformation, but note that if the stream contains inline parameter sets then the output may be unusable if they are removed.
For example, to remove all non-VCL NAL units from an H.264 stream:
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT
To remove all AUDs, SEI and filler from an H.265 stream:
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.
Specifies the texture to keep.
Convert HAPQA to HAPQ
ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov
Convert HAPQA to HAPAlphaOnly
ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov
Modify metadata embedded in an H.264 stream.
Insert or remove AUD NAL units in all access units of the stream.
Set the sample aspect ratio of the stream in the VUI parameters.
Set the video format in the stream (see H.264 section E.2.1 and table E-2).
Set the colour description in the stream (see H.264 section E.2.1 and tables E-3, E-4 and E-5).
Set the chroma sample location in the stream (see H.264 section E.2.1 and figure E-1).
Set the tick rate (num_units_in_tick / time_scale) in the VUI parameters. This is the smallest time unit representable in the stream, and in many cases represents the field rate of the stream (double the frame rate).
Set whether the stream has fixed framerate - typically this indicates that the framerate is exactly half the tick rate, but the exact meaning is dependent on interlacing and the picture structure (see H.264 section E.2.1 and table E-6).
Set the frame cropping offsets in the SPS. These values will replace the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be representable if the chroma is subsampled or the stream is interlaced (see H.264 section 7.4.2.1.1).
Insert a string as SEI unregistered user data. The argument must be of the form UUID+string, where the UUID is as hex digits possibly separated by hyphens, and the string can be anything.
For example, ‘086f3693-b7b3-4f2c-9653-21492feee5b8+hello’ will insert the string “hello” associated with the given UUID.
Deletes both filler NAL units and filler SEI messages.
Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.264 specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format (muxer mpegts).
For example to remux an MP4 file containing an H.264 stream to mpegts
format with ffmpeg, you can use the command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
Please note that this filter is auto-inserted for MPEG-TS (muxer
mpegts) and raw H.264 (muxer h264) output formats.
This applies a specific fixup to some Blu-ray streams which contain redundant PPSs modifying irrelevant parameters of the stream which confuse other transformations which require correct extradata.
A new single global PPS is created, and all of the redundant PPSs within the stream are removed.
Modify metadata embedded in an HEVC stream.
Insert or remove AUD NAL units in all access units of the stream.
Set the sample aspect ratio in the stream in the VUI parameters.
Set the video format in the stream (see H.265 section E.3.1 and table E.2).
Set the colour description in the stream (see H.265 section E.3.1 and tables E.3, E.4 and E.5).
Set the chroma sample location in the stream (see H.265 section E.3.1 and figure E.1).
Set the tick rate in the VPS and VUI parameters (num_units_in_tick / time_scale). Combined with ‘num_ticks_poc_diff_one’, this can set a constant framerate in the stream. Note that it is likely to be overridden by container parameters when the stream is in a container.
Set poc_proportional_to_timing_flag in VPS and VUI and use this value to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and E.3.1). Ignored if ‘tick_rate’ is not also set.
Set the conformance window cropping offsets in the SPS. These values will replace the current ones if the stream is already cropped.
These fields are set in pixels. Note that some sizes may not be representable if the chroma is subsampled (H.265 section 7.4.3.2.1).
Convert an HEVC/H.265 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.265 specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format (muxer mpegts).
For example to remux an MP4 file containing an HEVC stream to mpegts
format with ffmpeg, you can use the command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
Please note that this filter is auto-inserted for MPEG-TS (muxer
mpegts) and raw HEVC/H.265 (muxer h265 or
hevc) output formats.
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
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
Add an MJPEG A header to the bitstream, to enable decoding by Quicktime.
Extract a representable text file from MOV subtitles, stripping the metadata header from each subtitle packet.
See also the text2movsub filter.
Decompress non-standard compressed MP3 audio headers.
Modify metadata embedded in an MPEG-2 stream.
Set the display aspect ratio in the stream.
The following fixed values are supported:
Any other value will result in square pixels being signalled instead (see H.262 section 6.3.3 and table 6-3).
Set the frame rate in the stream. This is constructed from a table of known values combined with a small multiplier and divisor - if the supplied value is not exactly representable, the nearest representable value will be used instead (see H.262 section 6.3.3 and table 6-4).
Set the video format in the stream (see H.262 section 6.3.6 and table 6-6).
Set the colour description in the stream (see H.262 section 6.3.6 and tables 6-7, 6-8 and 6-9).
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
Damages the contents of packets or simply drops them 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.
A numeral string, whose value is related to how often packets will be dropped. Therefore, values below or equal to 0 are forbidden, and the lower the more frequent packets will be dropped, with 1 meaning every packet is dropped.
The following example applies the modification to every byte but does not drop any packets.
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
This bitstream filter passes the packets through unchanged.
Remove extradata from packets.
It accepts the following parameter:
Set which frame types to remove extradata from.
Remove extradata from non-keyframes only.
Remove extradata from keyframes only.
Remove extradata from all frames.
Convert text subtitles to MOV subtitles (as used by the mov_text
codec) with metadata headers.
See also the mov2textsub filter.
Log trace output containing all syntax elements in the coded stream headers (everything above the level of individual coded blocks). This can be useful for debugging low-level stream issues.
Supports H.264, H.265 and MPEG-2.
Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This fixes merging of split/segmented VP9 streams where the alt-ref frame was split from its visible counterpart.
Split VP9 superframes into single frames.
Given a VP9 stream with correct timestamps but possibly out of order, insert additional show-existing-frame packets to correct the ordering.
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:
Possible values:
Reduce buffering.
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.
Set packet size.
Set format flags.
Possible values:
Ignore index.
Enable fast, but inaccurate seeks for some formats.
Generate PTS.
Do not fill in missing values that can be exactly calculated.
Disable AVParsers, this needs +nofillin too.
Ignore DTS.
Discard corrupted frames.
Try to interleave output packets by DTS.
Do not merge side data.
Enable RTP MP4A-LATM payload.
Reduce the latency introduced by optional buffering
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.
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.
Allow seeking to non-keyframes on demuxer level when supported if set to 1. Default is 0.
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.
Set decryption key.
Set max memory used for timestamp index (per stream).
Set max memory used for buffering real-time frames.
Print specific debug info.
Possible values:
Set maximum muxing or demuxing delay in microseconds.
Set number of frames used to probe fps.
Set microseconds by which audio packets should be interleaved earlier.
Set microseconds for each chunk.
Set size in bytes for each chunk.
Set error detection flags. f_err_detect is deprecated and
should be used only via the ffmpeg tool.
Possible values:
Verify embedded CRCs.
Detect bitstream specification deviations.
Detect improper bitstream length.
Abort decoding on minor error detection.
Consider things that violate the spec and have not been seen in the wild as errors.
Consider all spec non compliancies as errors.
Consider things that a sane encoder should not do as an error.
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 if set to 1. Default is 0.
Possible values:
Shift timestamps to make them non-negative. Also note that this affects only leading negative timestamps, and not non-monotonic negative timestamps.
Shift timestamps so that the first timestamp is 0.
Enables shifting when required by the target format.
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.
Set number of bytes to skip before reading header and frames if set to 1. Default is 0.
Correct single timestamp overflows if set to 1. Default is 1.
Flush the underlying I/O stream after each packet. Default is -1 (auto), which means that the underlying protocol will decide, 1 enables it, and has the effect of reducing the latency, 0 disables it and may increase IO throughput in some cases.
Set the output time offset.
offset must be a time duration specification, see (ffmpeg-utils)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).
"," separated list of allowed demuxers. By default all are allowed.
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
Specifies the maximum number of streams. This can be used to reject files that would require too many resources due to a large number of streams.
Format stream specifiers allow selection of one or more streams that match specific properties.
Possible forms of stream specifiers are:
Matches the stream with this 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.
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.
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 are configured elements in FFmpeg that can read the multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers
are enabled by default. You can list all available ones using the
configure option --list-demuxers.
You can disable all the demuxers using the configure option
--disable-demuxers, and selectively enable a single demuxer with
the option --enable-demuxer=DEMUXER, or disable it
with the option --disable-demuxer=DEMUXER.
The option -demuxers of the ff* tools will display the list of
enabled demuxers. Use -formats to view a combined list of
enabled demuxers and muxers.
The description of some of the currently available demuxers follows.
Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
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".
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 the loop variable in the file if set.
Maximum framerate in frames per second (0 for no limit).
Default framerate in frames per second when none is specified in the file (0 meaning as fast as possible).
Advanced Systems Format demuxer.
This demuxer is used to demux ASF files and MMS network streams.
Do not try to resynchronize by looking for a certain optional start code.
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.
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.
This demuxer accepts the following option:
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.
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.
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.
# 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'
ffconcat version 1.0 file file-1.wav duration 20.0 file subdir/file-2.wav
Dynamic Adaptive Streaming over HTTP demuxer.
This demuxer presents all AVStreams found in the manifest.
By setting the discard flags on AVStreams the caller can decide
which streams to actually receive.
Each stream mirrors the id and bandwidth properties from the
<Representation> as metadata keys named "id" and "variant_bitrate" respectively.
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities.
ffmpeg -f flv -i myfile.flv ... ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
Allocate the streams according to the onMetaData array content.
Animated GIF demuxer.
It accepts the following options:
Set the minimum valid delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 2.
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.
Set the default delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 10.
GIF files can contain information to loop a certain number of times (or infinitely). If ‘ignore_loop’ is set to 1, then the loop setting from the input will be ignored and looping will not occur. If set to 0, then looping will occur and will cycle the number of times according to the GIF. Default value is 1.
For example, with the overlay filter, place an infinitely looping GIF over another video:
ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
Note that in the above example the shortest option for overlay filter is used to end the output video at the length of the shortest input file, which in this case is ‘input.mp4’ as the GIF in this example loops infinitely.
HLS demuxer
It accepts the following options:
segment index to start live streams at (negative values are from the end).
’,’ separated list of file extensions that hls is allowed to access.
Maximum number of times a insufficient list is attempted to be reloaded. Default value is 1000.
Use persistent HTTP connections. Applicable only for HTTP streams. Enabled by default.
Use multiple HTTP connections for downloading HTTP segments. Enabled by default for HTTP/1.1 servers.
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:
Set the frame rate for the video stream. It defaults to 25.
If set to 1, loop over the input. Default value is 0.
Select the pattern type used to interpret the provided filename.
pattern_type accepts one of the following values.
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.
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
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.
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.
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.
Set the index of the file matched by the image file pattern to start to read from. Default value is 0.
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.
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.
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.
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
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
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 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:
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.
Set the channel layout. Valid values are 1, 2, and 4 channel layouts. The default value is STEREO.
Set the sample rate for libopenmpt to output. Range is from 1000 to INT_MAX. The value default is 48000.
QuickTime / MP4 demuxer.
This demuxer accepts the following options:
Enable loading of external tracks, disabled by default. Enabling this can theoretically leak information in some use cases.
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.
MPEG-2 transport stream demuxer.
This demuxer accepts the following options:
Set size limit for looking up a new synchronization. Default value is 65536.
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.
Output option carrying the raw packet size in bytes. Show the detected raw packet size, cannot be set by the user.
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.
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.
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.
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:
Set input video frame rate. Default value is 25.
Set the input video pixel format. Default value is yuv420p.
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
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.
JSON captions used for TED Talks.
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:
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 are configured elements in FFmpeg which allow writing multimedia streams to a particular type of file.
When you configure your FFmpeg build, all the supported muxers
are enabled by default. You can list all available muxers using the
configure option --list-muxers.
You can disable all the muxers with the configure option
--disable-muxers and selectively enable / disable single muxers
with the options --enable-muxer=MUXER /
--disable-muxer=MUXER.
The option -muxers of the ff* tools will display the list of
enabled muxers. Use -formats to view a combined list of
enabled demuxers and muxers.
A description of some of the currently available muxers follows.
Audio Interchange File Format muxer.
It accepts the following options:
Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
Select ID3v2 version to write. Currently only version 3 and 4 (aka. ID3v2.3 and ID3v2.4) are supported. The default is version 4.
Advanced Systems Format muxer.
Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this muxer too.
It accepts the following options:
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.
Audio Video Interleaved muxer.
It accepts the following options:
Reserve the specified amount of bytes for the OpenDML master index of each stream within the file header. By default additional master indexes are embedded within the data packets if there is no space left in the first master index and are linked together as a chain of indexes. This index structure can cause problems for some use cases, e.g. third-party software strictly relying on the OpenDML index specification or when file seeking is slow. Reserving enough index space in the file header avoids these problems.
The required index space depends on the output file size and should be about 16 bytes per gigabyte. When this option is omitted or set to zero the necessary index space is guessed.
Write the channel layout mask into the audio stream header.
This option is enabled by default. Disabling the channel mask can be useful in specific scenarios, e.g. when merging multiple audio streams into one for compatibility with software that only supports a single audio stream in AVI (see (ffmpeg-filters)the "amerge" section in the ffmpeg-filters manual).
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.
Threshold for detecting silence, ranges from 0 to 32767. -1 for default (required for use with the AcoustID service).
Algorithm index to fingerprint with.
Format to output the fingerprint as. Accepts the following options:
Binary raw fingerprint
Binary compressed fingerprint
Base64 compressed fingerprint
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.
The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames.
See also the framecrc muxer.
For example to compute the CRC of the input, and store it in the file ‘out.crc’:
ffmpeg -i INPUT -f crc out.crc
You can print the CRC to stdout with the command:
ffmpeg -i INPUT -f crc -
You can select the output format of each frame with ffmpeg by
specifying the audio and video codec and format. For example to
compute the CRC of the input audio converted to PCM unsigned 8-bit
and the input video converted to MPEG-2 video, use the command:
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
Adobe Flash Video Format muxer.
This muxer accepts the following options:
Possible values:
Place AAC sequence header based on audio stream data.
Disable sequence end tag.
Disable metadata tag.
Disable duration and filesize in metadata when they are equal to zero at the end of stream. (Be used to non-seekable living stream).
Used to facilitate seeking; particularly for HTTP pseudo streaming.
Dynamic Adaptive Streaming over HTTP (DASH) muxer that creates segments and manifest files according to the MPEG-DASH standard ISO/IEC 23009-1:2014.
For more information see:
It creates a MPD manifest file and segment files for each stream.
The segment filename might contain pre-defined identifiers used with SegmentTemplate as defined in section 5.3.9.4.4 of the standard. Available identifiers are "$RepresentationID$", "$Number$", "$Bandwidth$" and "$Time$".
ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264 -b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline -profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0 -b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1 -window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a" -f dash /path/to/out.mpd
Set the segment length in microseconds.
Set the maximum number of segments kept in the manifest.
Set the maximum number of segments kept outside of the manifest before removing from disk.
Enable (1) or disable (0) removal of all segments when finished.
Enable (1) or disable (0) use of SegmentTemplate instead of SegmentList.
Enable (1) or disable (0) use of SegmentTimeline in SegmentTemplate.
Enable (1) or disable (0) storing all segments in one file, accessed using byte ranges.
DASH-templated name to be used for baseURL. Implies single_file set to "1".
DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.m4s"
DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.m4s"
URL of the page that will return the UTC timestamp in ISO format. Example: "https://time.akamai.com/?iso"
Override User-Agent field in HTTP header. Applicable only for HTTP output.
Use persistent HTTP connections. Applicable only for HTTP output.
Generate HLS playlist files as well. The master playlist is generated with the filename master.m3u8. One media playlist file is generated for each stream with filenames media_0.m3u8, media_1.m3u8, etc.
Enable (1) or disable (0) chunk streaming mode of output. In chunk streaming mode, each frame will be a moof fragment which forms a chunk.
Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c id=y,streams=d,e" with x and y being the IDs of the adaptation sets and a,b,c,d and e are the indices of the mapped streams.
To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be used as stream identifier instead of IDs.
When no assignment is defined, this defaults to an AdaptationSet for each stream.
Set timeout for socket I/O operations. Applicable only for HTTP output.
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, 0xCRC
CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the packet.
For example to compute the CRC of the audio and video frames in ‘INPUT’, converted to raw audio and video packets, and store it in the file ‘out.crc’:
ffmpeg -i INPUT -f framecrc out.crc
To print the information to stdout, use the command:
ffmpeg -i INPUT -f framecrc -
With ffmpeg, you can select the output format to which the
audio and video frames are encoded before computing the CRC for each
packet by specifying the audio and video codec. For example, to
compute the CRC of each decoded input audio frame converted to PCM
unsigned 8-bit and of each decoded input video frame converted to
MPEG-2 video, use the command:
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
See also the crc muxer.
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.
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.
To compute the SHA-256 hash of the audio and video frames in ‘INPUT’, converted to raw audio and video packets, and store it in the file ‘out.sha256’:
ffmpeg -i INPUT -f framehash out.sha256
To print the information to stdout, using the MD5 hash function, use the command:
ffmpeg -i INPUT -f framehash -hash md5 -
See also the hash muxer.
Per-packet MD5 testing format.
This is a variant of the framehash muxer. Unlike that muxer, it defaults to using the MD5 hash function.
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.
Animated GIF muxer.
It accepts the following options:
Set the number of times to loop the output. Use -1 for no loop, 0
for looping indefinitely (default).
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 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.
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.
To compute the SHA-256 hash of the input converted to raw audio and video, and store it in the file ‘out.sha256’:
ffmpeg -i INPUT -f hash out.sha256
To print an MD5 hash to stdout use the command:
ffmpeg -i INPUT -f hash -hash md5 -
See also the framehash muxer.
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.
Make sure to require a closed GOP when encoding and to set the GOP size to fit your segment time constraint.
For example, to convert an input file with ffmpeg:
ffmpeg -i in.mkv -c:v h264 -flags +cgop -g 30 -hls_time 1 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.
This muxer supports the following options:
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
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.
Set the maximum number of playlist entries. If set to 0 the list file will contain all the segments. Default value is 5.
Set the number of unreferenced segments to keep on disk before hls_flags delete_segments
deletes them. Increase this to allow continue clients to download segments which
were recently referenced in the playlist. Default value is 1, meaning segments older than
hls_list_size+1 will be deleted.
Set output format options using a :-separated list of key=value
parameters. Values containing : special characters must be
escaped.
This is a deprecated option, you can use hls_list_size
and hls_flags delete_segments instead it
This option is useful to avoid to fill the disk with many segment files, and limits the maximum number of segment files written to disk to wrap.
Start the playlist sequence number (#EXT-X-MEDIA-SEQUENCE) according to the specified source.
Unless hls_flags single_file is set, it also specifies source of starting sequence numbers of
segment and subtitle filenames. In any case, if hls_flags append_list
is set and read playlist sequence number is greater than the specified start sequence number,
then that value will be used as start value.
It accepts the following values:
Set the starting sequence numbers according to start_number option value.
The start number will be the seconds since epoch (1970-01-01 00:00:00)
The start number will be based on the current date/time as YYYYmmddHHMMSS. e.g. 20161231235759.
Start the playlist sequence number (#EXT-X-MEDIA-SEQUENCE) from the specified number
when hls_start_number_source value is generic. (This is the default case.)
Unless hls_flags single_file is set, it also specifies starting sequence numbers of segment and subtitle filenames.
Default value is 0.
Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments.
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.
Set the segment filename. Unless hls_flags single_file is set,
filename is used as a string format with the segment number:
ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8
This example will produce the playlist, ‘out.m3u8’, and segment files: ‘file000.ts’, ‘file001.ts’, ‘file002.ts’, etc.
filename may contain full path or relative path specification, but only the file name part without any path info will be contained in the m3u8 segment list. Should a relative path be specified, the path of the created segment files will be relative to the current working directory. When use_localtime_mkdir is set, the whole expanded value of filename will be written into the m3u8 segment list.
When var_stream_map is set with two or more variant streams, the
filename pattern must contain the string "%v", this string specifies
the position of variant stream index in the generated segment file names.
ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \ -hls_segment_filename 'file_%v_%03d.ts' out_%v.m3u8
This example will produce the playlists segment file sets: ‘file_0_000.ts’, ‘file_0_001.ts’, ‘file_0_002.ts’, etc. and ‘file_1_000.ts’, ‘file_1_001.ts’, ‘file_1_002.ts’, etc.
The string "%v" may be present in the filename or in the last directory name containing the file. If the string is present in the directory name, then sub-directories are created after expanding the directory name pattern. This enables creation of segments corresponding to different variant streams in subdirectories.
ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \ -hls_segment_filename 'vs%v/file_%03d.ts' vs%v/out.m3u8
This example will produce the playlists segment file sets: ‘vs0/file_000.ts’, ‘vs0/file_001.ts’, ‘vs0/file_002.ts’, etc. and ‘vs1/file_000.ts’, ‘vs1/file_001.ts’, ‘vs1/file_002.ts’, etc.
Use strftime() on filename to expand the segment filename with localtime. The segment number is also available in this mode, but to use it, you need to specify second_level_segment_index hls_flag and %%d will be the specifier.
ffmpeg -i in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
This example will produce the playlist, ‘out.m3u8’, and segment files:
‘file-20160215-1455569023.ts’, ‘file-20160215-1455569024.ts’, etc.
Note: On some systems/environments, the %s specifier is not available. See
strftime() documentation.
ffmpeg -i in.nut -use_localtime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8
This example will produce the playlist, ‘out.m3u8’, and segment files: ‘file-20160215-0001.ts’, ‘file-20160215-0002.ts’, etc.
Used together with -use_localtime, it will create all subdirectories which is expanded in filename.
ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
This example will create a directory 201560215 (if it does not exist), and then produce the playlist, ‘out.m3u8’, and segment files: ‘20160215/file-20160215-1455569023.ts’, ‘20160215/file-20160215-1455569024.ts’, etc.
ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8
This example will create a directory hierarchy 2016/02/15 (if any of them do not exist), and then produce the playlist, ‘out.m3u8’, and segment files: ‘2016/02/15/file-20160215-1455569023.ts’, ‘2016/02/15/file-20160215-1455569024.ts’, etc.
Use the information in key_info_file for segment encryption. The first
line of key_info_file specifies the key URI written to the playlist. The
key URL is used to access the encryption key during playback. The second line
specifies the path to the key file used to obtain the key during the encryption
process. The key file is read as a single packed array of 16 octets in binary
format. The optional third line specifies the initialization vector (IV) as a
hexadecimal string to be used instead of the segment sequence number (default)
for encryption. Changes to key_info_file will result in segment
encryption with the new key/IV and an entry in the playlist for the new key
URI/IV if hls_flags periodic_rekey is enabled.
Key info file format:
key URI key file path IV (optional)
Example key URIs:
http://server/file.key /path/to/file.key file.key
Example key file paths:
file.key /path/to/file.key
Example IV:
0123456789ABCDEF0123456789ABCDEF
Key info file example:
http://server/file.key /path/to/file.key 0123456789ABCDEF0123456789ABCDEF
Example shell script:
#!/bin/sh
BASE_URL=${1:-'.'}
openssl rand 16 > file.key
echo $BASE_URL/file.key > file.keyinfo
echo file.key >> file.keyinfo
echo $(openssl rand -hex 16) >> file.keyinfo
ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
-hls_key_info_file file.keyinfo out.m3u8
Enable (1) or disable (0) the AES128 encryption. When enabled every segment generated is encrypted and the encryption key is saved as playlist name.key.
Hex-coded 16byte key to encrypt the segments, by default it is randomly generated.
If set, keyurl is prepended instead of baseurl to the key filename in the playlist.
Hex-coded 16byte initialization vector for every segment instead of the autogenerated ones.
Possible values:
If this flag is set, the hls segment files will format to mpegts. the mpegts files is used in all hls versions.
If this flag is set, the hls segment files will format to fragment mp4 looks like dash. the fmp4 files is used in hls after version 7.
set filename to the fragment files header file, default filename is ‘init.mp4’.
When var_stream_map is set with two or more variant streams, the
filename pattern must contain the string "%v", this string specifies
the position of variant stream index in the generated init file names.
The string "%v" may be present in the filename or in the last directory name
containing the file. If the string is present in the directory name, then
sub-directories are created after expanding the directory name pattern. This
enables creation of init files corresponding to different variant streams in
subdirectories.
Possible values:
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’.
Segment files removed from the playlist are deleted after a period of time equal to the duration of the segment plus the duration of the playlist.
Append new segments into the end of old segment list,
and remove the #EXT-X-ENDLIST from the old segment list.
Round the duration info in the playlist file segment info to integer values, instead of using floating point.
Add the #EXT-X-DISCONTINUITY tag to the playlist, before the
first segment’s information.
Do not append the EXT-X-ENDLIST tag at the end of the playlist.
The file specified by hls_key_info_file will be checked periodically and
detect updates to the encryption info. Be sure to replace this file atomically,
including the file containing the AES encryption key.
Add the #EXT-X-INDEPENDENT-SEGMENTS to playlists that has video segments
and when all the segments of that playlist are guaranteed to start with a Key frame.
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.
Generate EXT-X-PROGRAM-DATE-TIME tags.
Makes it possible to use segment indexes as %%d in hls_segment_filename expression besides date/time values when use_localtime is on. To get fixed width numbers with trailing zeroes, %%0xd format is available where x is the required width.
Makes it possible to use segment sizes (counted in bytes) as %%s in hls_segment_filename expression besides date/time values when use_localtime is on. To get fixed width numbers with trailing zeroes, %%0xs format is available where x is the required width.
Makes it possible to use segment duration (calculated in microseconds) as %%t in hls_segment_filename expression besides date/time values when use_localtime is on. To get fixed width numbers with trailing zeroes, %%0xt format is available where x is the required width.
ffmpeg -i sample.mpeg \ -f hls -hls_time 3 -hls_list_size 5 \ -hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \ -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
This will produce segments like this: ‘segment_20170102194334_0003_00122200_0000003000000.ts’, ‘segment_20170102194334_0004_00120072_0000003000000.ts’ etc.
Write segment data to filename.tmp and rename to filename only once the segment is complete. A webserver serving up segments can be configured to reject requests to *.tmp to prevent access to in-progress segments before they have been added to the m3u8 playlist.
Emit #EXT-X-PLAYLIST-TYPE:EVENT in the m3u8 header. Forces
‘hls_list_size’ to 0; the playlist can only be appended to.
Emit #EXT-X-PLAYLIST-TYPE:VOD in the m3u8 header. Forces
‘hls_list_size’ to 0; the playlist must not change.
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.
Override User-Agent field in HTTP header. Applicable only for HTTP output.
Map string which specifies how to group the audio, video and subtitle streams into different variant streams. The variant stream groups are separated by space. Expected string format is like this "a:0,v:0 a:1,v:1 ....". Here a:, v:, s: are the keys to specify audio, video and subtitle streams respectively. Allowed values are 0 to 9 (limited just based on practical usage).
When there are two or more variant streams, the output filename pattern must contain the string "%v", this string specifies the position of variant stream index in the output media playlist filenames. The string "%v" may be present in the filename or in the last directory name containing the file. If the string is present in the directory name, then sub-directories are created after expanding the directory name pattern. This enables creation of variant streams in subdirectories.
ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \ http://example.com/live/out_%v.m3u8
This example creates two hls variant streams. The first variant stream will contain video stream of bitrate 1000k and audio stream of bitrate 64k and the second variant stream will contain video stream of bitrate 256k and audio stream of bitrate 32k. Here, two media playlist with file names out_0.m3u8 and out_1.m3u8 will be created.
ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k \ -map 0:v -map 0:a -map 0:v -f hls -var_stream_map "v:0 a:0 v:1" \ http://example.com/live/out_%v.m3u8
This example creates three hls variant streams. The first variant stream will be a video only stream with video bitrate 1000k, the second variant stream will be an audio only stream with bitrate 64k and the third variant stream will be a video only stream with bitrate 256k. Here, three media playlist with file names out_0.m3u8, out_1.m3u8 and out_2.m3u8 will be created.
ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \ http://example.com/live/vs_%v/out.m3u8
This example creates the variant streams in subdirectories. Here, the first media playlist is created at ‘http://example.com/live/vs_0/out.m3u8’ and the second one at ‘http://example.com/live/vs_1/out.m3u8’.
ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k -b:v:1 3000k \ -map 0:a -map 0:a -map 0:v -map 0:v -f hls \ -var_stream_map "a:0,agroup:aud_low a:1,agroup:aud_high v:0,agroup:aud_low v:1,agroup:aud_high" \ -master_pl_name master.m3u8 \ http://example.com/live/out_%v.m3u8
This example creates two audio only and two video only variant streams. In addition to the #EXT-X-STREAM-INF tag for each variant stream in the master playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams and they are mapped to the two video only variant streams with audio group names ’aud_low’ and ’aud_high’.
By default, a single hls variant containing all the encoded streams is created.
Map string which specifies different closed captions groups and their
attributes. The closed captions stream groups are separated by space.
Expected string format is like this
"ccgroup:<group name>,instreamid:<INSTREAM-ID>,language:<language code> ....".
’ccgroup’ and ’instreamid’ are mandatory attributes. ’language’ is an optional
attribute.
The closed captions groups configured using this option are mapped to different
variant streams by providing the same ’ccgroup’ name in the
var_stream_map string. If var_stream_map is not set, then the
first available ccgroup in cc_stream_map is mapped to the output variant
stream. The examples for these two use cases are given below.
ffmpeg -re -i in.ts -b:v 1000k -b:a 64k -a53cc 1 -f hls \ -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en" \ -master_pl_name master.m3u8 \ http://example.com/live/out.m3u8
This example adds #EXT-X-MEDIA tag with TYPE=CLOSED-CAPTIONS in
the master playlist with group name ’cc’, langauge ’en’ (english) and
INSTREAM-ID ’CC1’. Also, it adds CLOSED-CAPTIONS attribute with group
name ’cc’ for the output variant stream.
ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -a53cc:0 1 -a53cc:1 1\ -map 0:v -map 0:a -map 0:v -map 0:a -f hls \ -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en ccgroup:cc,instreamid:CC2,language:sp" \ -var_stream_map "v:0,a:0,ccgroup:cc v:1,a:1,ccgroup:cc" \ -master_pl_name master.m3u8 \ http://example.com/live/out_%v.m3u8
This example adds two #EXT-X-MEDIA tags with TYPE=CLOSED-CAPTIONS in
the master playlist for the INSTREAM-IDs ’CC1’ and ’CC2’. Also, it adds
CLOSED-CAPTIONS attribute with group name ’cc’ for the two output variant
streams.
Create HLS master playlist with the given name.
ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 http://example.com/live/out.m3u8
This example creates HLS master playlist with name master.m3u8 and it is published at http://example.com/live/
Publish master play list repeatedly every after specified number of segment intervals.
ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 \ -hls_time 2 -master_pl_publish_rate 30 http://example.com/live/out.m3u8
This example creates HLS master playlist with name master.m3u8 and keep publishing it repeatedly every after 30 segments i.e. every after 60s.
Use persistent HTTP connections. Applicable only for HTTP output.
Set timeout for socket I/O operations. Applicable only for HTTP output.
ICO file muxer.
Microsoft’s icon file format (ICO) has some strict limitations that should be noted:
BMP Bit Depth FFmpeg Pixel Format 1bit pal8 4bit pal8 8bit pal8 16bit rgb555le 24bit bgr24 32bit bgra
Image file muxer.
The image file muxer writes video frames to image files.
The output filenames are specified by a pattern, which can be used to produce sequentially numbered series of files. The pattern may contain the string "%d" or "%0Nd", this string specifies the position of the characters representing a numbering in the filenames. If the form "%0Nd" is used, the string representing the number in each filename is 0-padded to N digits. The literal character ’%’ can be specified in the pattern with the string "%%".
If the pattern contains "%d" or "%0Nd", the first filename of the file list specified will contain the number 1, all the following numbers will be sequential.
The pattern may contain a suffix which is used to automatically determine the format of the image files to write.
For example the pattern "img-%03d.bmp" will specify a sequence of filenames of the form ‘img-001.bmp’, ‘img-002.bmp’, ..., ‘img-010.bmp’, etc. The pattern "img%%-%d.jpg" will specify a sequence of filenames of the form ‘img%-1.jpg’, ‘img%-2.jpg’, ..., ‘img%-10.jpg’, etc.
The following example shows how to use ffmpeg for creating a
sequence of files ‘img-001.jpeg’, ‘img-002.jpeg’, ...,
taking one image every second from the input video:
ffmpeg -i in.avi -vsync cfr -r 1 -f image2 'img-%03d.jpeg'
Note that with ffmpeg, if the format is not specified with the
-f option and the output filename specifies an image file
format, the image2 muxer is automatically selected, so the previous
command can be written as:
ffmpeg -i in.avi -vsync cfr -r 1 'img-%03d.jpeg'
Note also that the pattern must not necessarily contain "%d" or "%0Nd", for example to create a single image file ‘img.jpeg’ from the start of the input video you can employ the command:
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
The ‘strftime’ option allows you to expand the filename with
date and time information. Check the documentation of
the strftime() function for the syntax.
For example to generate image files from the strftime()
"%Y-%m-%d_%H-%M-%S" pattern, the following ffmpeg command
can be used:
ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
You can set the file name with current frame’s PTS:
ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg"
If set to 1, expand the filename with pts from pkt->pts. Default value is 0.
Start the sequence from the specified number. Default value is 1.
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.
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 container muxer.
This muxer implements the matroska and webm container specs.
The recognized metadata settings in this muxer are:
Set title name provided to a single track.
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).
Set stereo 3D video layout of two views in a single video track.
The following values are recognized:
video is not stereo
Both views are arranged side by side, Left-eye view is on the left
Both views are arranged in top-bottom orientation, Left-eye view is at bottom
Both views are arranged in top-bottom orientation, Left-eye view is on top
Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
Each view is constituted by a row based interleaving, Right-eye view is first row
Each view is constituted by a row based interleaving, Left-eye view is first row
Both views are arranged in a column based interleaving manner, Right-eye view is first column
Both views are arranged in a column based interleaving manner, Left-eye view is first column
All frames are in anaglyph format viewable through red-cyan filters
Both views are arranged side by side, Right-eye view is on the left
All frames are in anaglyph format viewable through green-magenta filters
Both eyes laced in one Block, Left-eye view is first
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
This muxer supports the following options:
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 testing format.
This is a variant of the hash muxer. Unlike that muxer, it defaults to using the MD5 hash function.
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 (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.
Fragmentation is enabled by setting one of the AVOptions that define how to cut the file into fragments:
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.
Start a new fragment at each video keyframe.
Create fragments that are duration microseconds long.
Create fragments that contain up to size bytes of payload data.
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.)
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:
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.
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.
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.
Add RTP hinting tracks to the output file.
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.
Do not write any absolute base_data_offset in tfhd atoms. This avoids tying fragments to absolute byte positions in the file/streams.
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).
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).
Enables utilization of version 1 of the CTTS box, in which the CTS offsets can be negative. This enables the initial sample to have DTS/CTS of zero, and reduces the need for edit lists for some cases such as video tracks with B-frames. Additionally, eases conformance with the DASH-IF interoperability guidelines.
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 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
The MP3 muxer writes a raw MP3 stream with the following optional features:
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.
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.
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
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’.
The muxer options are:
Set the ‘transport_stream_id’. This identifies a transponder in DVB.
Default is 0x0001.
Set the ‘original_network_id’. This is unique identifier of a
network in DVB. Its main use is in the unique identification of a service
through the path ‘Original_Network_ID, Transport_Stream_ID’. Default
is 0x0001.
Set the ‘service_id’, also known as program in DVB. Default is
0x0001.
Set the program ‘service_type’. Default is digital_tv.
Accepts the following options:
Any hexdecimal value between 0x01 to 0xff as defined in
ETSI 300 468.
Digital TV service.
Digital Radio service.
Teletext service.
Advanced Codec Digital Radio service.
MPEG2 Digital HDTV service.
Advanced Codec Digital SDTV service.
Advanced Codec Digital HDTV service.
Set the first PID for PMT. Default is 0x1000. Max is 0x1f00.
Set the first PID for data packets. Default is 0x0100. Max is
0x0f00.
Enable m2ts mode if set to 1. Default value is -1 which
disables m2ts mode.
Set a constant muxrate. Default is VBR.
Set minimum PES packet payload in bytes. Default is 2930.
Set mpegts flags. Accepts the following options:
Reemit PAT/PMT before writing the next packet.
Use LATM packetization for AAC.
Reemit PAT and PMT at each video frame.
Conform to System B (DVB) instead of System A (ATSC).
Mark the initial packet of each stream as discontinuity.
Reemit PAT/PMT before writing the next packet. This option is deprecated: use ‘mpegts_flags’ instead.
Preserve original timestamps, if value is set to 1. Default value
is -1, which results in shifting timestamps so that they start from 0.
Omit the PES packet length for video packets. Default is 1 (true).
Override the default PCR retransmission time in milliseconds. Ignored if
variable muxrate is selected. Default is 20.
Maximum time in seconds between PAT/PMT tables.
Maximum time in seconds between SDT tables.
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 ...
ffmpeg -i file.mpg -c copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
out.ts
MXF muxer.
The muxer options are:
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 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 -
Change the syncpoint usage in nut:
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, -write_index 0 can be used to disable
all growing data tables, allowing to mux endless streams with limited memory
and without these disadvantages.
The none and timestamped flags are experimental.
Write index at the end, the default is to write an index.
ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor
Ogg container muxer.
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 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.
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.
The segment muxer supports the following options:
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.
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.
Override the inner container format, by default it is guessed by the filename extension.
Set output format options using a :-separated list of key=value
parameters. Values containing the : special character must be
escaped.
Generate also a listfile named name. If not specified no listfile is generated.
Set flags affecting the segment list generation.
It currently supports the following flags:
Allow caching (only affects M3U8 list files).
Allow live-friendly file generation.
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.
Prepend prefix to each entry. Useful to generate absolute paths. By default no prefix is applied.
Select the listing format.
The following values are recognized:
Generate a flat list for the created segments, one segment per line.
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’.
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.
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.
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.
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".
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".
Force the segmenter to only start a new segment if a packet reaches the muxer within the specified duration after the segmenting clock time. This way you can make the segmenter more resilient to backward local time jumps, such as leap seconds or transition to standard time from daylight savings time.
Default is the maximum possible duration which means starting a new segment regardless of the elapsed time since the last clock time.
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.
Specify a list of split points. times contains a list of comma separated duration specifications, in increasing order. See also the ‘segment_time’ option.
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.
Wrap around segment index once it reaches limit.
Set the sequence number of the first segment. Defaults to 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.
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 at the beginning of each segment, so that each segment
will start with near-zero timestamps. It is meant to ease the playback
of the generated segments. May not work with some combinations of
muxers/codecs. It is set to 0 by default.
Specify timestamp offset to apply to the output packet timestamps. The argument must be a time duration specification, and defaults to 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.
Make sure to require a closed GOP when encoding and to set the GOP size to fit your segment time constraint.
ffmpeg -i in.mkv -codec hevc -flags +cgop -g 60 -map 0 -f segment -segment_list out.list out%03d.nut
ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4
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
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.
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut
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
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
Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving with conventional web server.
Specify the number of fragments kept in the manifest. Default 0 (keep all).
Specify the number of fragments kept outside of the manifest before removing from disk. Default 5.
Specify the number of lookahead fragments. Default 2.
Specify the minimum fragment duration (in microseconds). Default 5000000.
Specify whether to remove all fragments when finished. Default 0 (do not remove).
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,
Specify the format name. Useful if it cannot be guessed from the output name suffix.
Specify size of the queue (number of packets). Default value is 60.
Specify format options for the underlying muxer. Muxer options can be specified as a list of key=value pairs separated by ’:’.
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.
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).
Sets maximum number of successive unsuccessful recovery attempts after which the output fails permanently. By default this option is set to 0 (unlimited).
Waiting time before the next recovery attempt after previous unsuccessful recovery attempt. Default value is 5 seconds.
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).
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.
Specify whether to wait for the keyframe after recovering from queue overflow or failure. This option is set to 0 (false) by default.
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
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.
If set to 1, slave outputs will be processed in separate thread using fifo muxer. This allows to compensate for different speed/latency/reliability of outputs and setup transparent recovery. By default this feature is turned off.
Options to pass to fifo pseudo-muxer instances. See fifo.
The slave outputs are specified in the file name given to the muxer, separated by ’|’. If any of the slave name contains the ’|’ separator, leading or trailing spaces or any special character, it must be escaped (see (ffmpeg-utils)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:
Specify the format name. Useful if it cannot be guessed from the output name suffix.
Specify a list of bitstream filters to apply to the specified output.
This allows to override tee muxer use_fifo option for individual slave muxer.
This allows to override tee muxer fifo_options for individual slave muxer. See fifo.
It is possible to specify to which streams a given bitstream filter
applies, by appending a stream specifier to the option separated by
/. spec must be a stream specifier (see Format stream specifiers). If the stream specifier is not specified, the
bitstream filters will be applied to all streams in the output.
Several bitstream filters can be specified, separated by ",".
Select 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
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.
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/"
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/"
ffmpeg to encode the input, and send the output
to three different destinations. The dump_extra bitstream
filter is used to add extradata information to all the output video
keyframes packets, as requested by the MPEG-TS format. The select
option is applied to ‘out.aac’ in order to make it contain only
audio packets.
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
a:1 for the audio output. Note
that a second level escaping must be performed, as ":" is a special
character used to separate options.
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
Note: some codecs may need different options depending on the output format; the auto-detection of this can not work with the tee muxer. The main example is the ‘global_header’ flag.
WebM DASH Manifest 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:
This muxer supports the following options:
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.
Set this to 1 to create a live stream DASH Manifest. Default: 0.
Start index of the first chunk. This will go in the ‘startNumber’ attribute of the ‘SegmentTemplate’ element in the manifest. Default: 0.
Duration of each chunk in milliseconds. This will go in the ‘duration’ attribute of the ‘SegmentTemplate’ element in the manifest. Default: 1000.
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.
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 (in seconds) of the manifest. This will go in the ‘minimumUpdatePeriod’ attribute of the ‘MPD’ element. Default: 0.
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 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.
This muxer supports the following options:
Index of the first chunk (defaults to 0).
Filename of the header where the initialization data will be written.
Duration of each audio chunk in milliseconds (defaults to 5000).
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
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:
Next a chapter section must contain chapter start and end times in form ‘START=num’, ‘END=num’, where num is a positive integer.
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
The libavformat library provides some generic global options, which can be set on all the protocols. In addition each protocol may support so-called private options, which are specific for that component.
Options may be set by specifying -option value in the
FFmpeg tools, or by setting the value explicitly in the
AVFormatContext options or using the ‘libavutil/opt.h’ API
for programmatic use.
The list of supported options follows:
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 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:
Maximum time to wait for (network) read/write operations to complete, in microseconds.
A description of the currently available protocols follows.
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
Read BluRay playlist.
The accepted options are:
BluRay angle
Start chapter (1...N)
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
Caching wrapper for input stream.
Cache the input stream to temporary file. It brings seeking capability to live streams.
cache:URL
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.
AES-encrypted stream reading protocol.
The accepted options are:
Set the AES decryption key binary block from given hexadecimal representation.
Set the AES decryption initialization vector binary block from given hexadecimal representation.
Accepted URL formats:
crypto:URL crypto+URL
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 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 existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
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 (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.
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.
Password used when login as anonymous user. Typically an e-mail address should be used.
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:
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 protocol.
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 (Hyper Text Transfer Protocol).
This protocol accepts the following options:
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.
If set to 1 use chunked Transfer-Encoding for posts, default is 1.
Set a specific content type for the POST messages or for listen mode.
set HTTP proxy to tunnel through e.g. http://example.com:1234
Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers.
Use persistent connections if set to 1, default is 0.
Set custom HTTP post data.
Set the Referer header. Include ’Referer: URL’ header in HTTP request.
Override the User-Agent header. If not specified the protocol will use a string describing the libavformat build. ("Lavf/<version>")
This is a deprecated option, you can use user_agent instead it.
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.
If set then eof is treated like an error and causes reconnection, this is useful for live / endless streams.
If set then even streamed/non seekable streams will be reconnected on errors.
Sets the maximum delay in seconds after which to give up reconnecting
Export the MIME type.
Exports the HTTP response version number. Usually "1.0" or "1.1".
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.
If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers, separated by newline characters.
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.
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.
Set initial byte offset.
Try to limit the request to bytes preceding this offset.
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.
If set to 1 enables experimental HTTP server. This can be used to send data when used as an output option, or read data from a client with HTTP POST when used as an input option. If set to 2 enables experimental multi-client HTTP server. This is not yet implemented in ffmpeg.c and thus must not be used as a command line option.
# Server side (sending): ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://server:port # Client side (receiving): ffmpeg -i http://server:port -c copy somefile.ogg # Client can also be done with wget: wget http://server:port -O somefile.ogg # Server side (receiving): ffmpeg -listen 1 -i http://server:port -c copy somefile.ogg # Client side (sending): ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://server:port # Client can also be done with wget: wget --post-file=somefile.ogg http://server:port
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 protocol (stream to Icecast servers)
This protocol accepts the following options:
Set the stream genre.
Set the stream name.
Set the stream description.
Set the stream website URL.
Set if the stream should be public. The default is 0 (not public).
Override the User-Agent header. If not specified a string of the form "Lavf/<version>" will be used.
Set the Icecast mountpoint password.
Set the stream content type. This must be set if it is different from audio/mpeg.
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
MMS (Microsoft Media Server) protocol over TCP.
MMS (Microsoft Media Server) protocol over HTTP.
The required syntax is:
mmsh://server[:port][/app][/playpath]
MD5 output protocol.
Computes the MD5 hash of the data to be written, and on close writes this to the designated output or stdout if none is specified. It can be used to test muxers without writing an actual file.
Some examples follow.
# Write the MD5 hash of the encoded AVI file to the file output.avi.md5. ffmpeg -i input.flv -f avi -y md5:output.avi.md5 # Write the MD5 hash of the encoded AVI file to stdout. ffmpeg -i input.flv -f avi -y md5:
Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with the MD5 output protocol.
UNIX pipe access protocol.
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:
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.
Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism for MPEG-2 Transport Streams sent over RTP.
This protocol must be used in conjunction with the rtp_mpegts muxer and
the rtp protocol.
The required syntax is:
-f rtp_mpegts -fec prompeg=option=val... rtp://hostname:port
The destination UDP ports are port + 2 for the column FEC stream
and port + 4 for the row FEC stream.
This protocol accepts the following options:
The number of columns (4-20, LxD <= 100)
The number of rows (4-20, LxD <= 100)
Example usage:
-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://hostname:port
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:
An optional username (mostly for publishing).
An optional password (mostly for publishing).
The address of the RTMP server.
The number of the TCP port to use (by default is 1935).
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.
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.
Act as a server, listening for an incoming connection.
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 AVOptions):
Name of application to connect on the RTMP server. This option overrides the parameter specified in the URI.
Set the client buffer time in milliseconds. The default is 3000.
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.
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>).)
Number of packets flushed in the same request (RTMPT only). The default is 10.
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.
URL of the web page in which the media was embedded. By default no value will be sent.
Stream identifier to play or to publish. This option overrides the parameter specified in the URI.
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.
SHA256 hash of the decompressed SWF file (32 bytes).
Size of the decompressed SWF file, required for SWFVerification.
URL of the SWF player for the media. By default no value will be sent.
URL to player swf file, compute hash/size automatically.
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/
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.
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.
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.
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.
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 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.
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 existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
Set the workgroup used for making connections. By default workgroup is not specified.
For more information see: http://www.samba.org/.
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.
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 existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
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
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"
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:
Set the TTL (Time-To-Live) value (for multicast only).
Set the remote RTCP port to n.
Set the local RTP port to n.
Set the local RTCP port to n.
Set max packet size (in bytes) to n.
Do a connect() on the UDP socket (if set to 1) or not (if set
to 0).
List allowed source IP addresses.
List disallowed (blocked) source IP addresses.
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).
Set the local RTP port to n.
This is a deprecated option. Instead, ‘localrtpport’ should be used.
Important notes:
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 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 AVOptions or in
avformat_open_input.
The following options are supported.
Do not start playing the stream immediately if set to 1. Default value is 0.
Set RTSP transport protocols.
It accepts the following values:
Use UDP as lower transport protocol.
Use TCP (interleaving within the RTSP control channel) as lower transport protocol.
Use UDP multicast as lower transport protocol.
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.
Set RTSP flags.
The following values are accepted:
Accept packets only from negotiated peer address and port.
Act as a server, listening for an incoming connection.
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
Default value is ‘none’.
Set media types to accept from the server.
The following flags are accepted:
By default it accepts all media types.
Set minimum local UDP port. Default value is 5000.
Set maximum local UDP port. Default value is 65000.
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’.
Set number of packets to buffer for handling of reordered packets.
Set socket TCP I/O timeout in microseconds.
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.
The following examples all make use of the ffplay and
ffmpeg tools.
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
ffplay -rtsp_transport http rtsp://server/video.mp4
ffmpeg -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp output
Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in libavformat, it is a muxer and demuxer. It is used for signalling of RTP streams, by announcing the SDP for the streams regularly on a separate port.
The syntax for a SAP url given to the muxer is:
sap://destination[:port][?options]
The RTP packets are sent to destination on port port,
or to port 5004 if no port is specified.
options is a &-separated list. The following options
are supported:
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.
Specify the port to send the announcements on, defaults to 9875 if not specified.
Specify the time to live value for the announcements and RTP packets, defaults to 255.
If set to 1, send all RTP streams on the same port pair. If zero (the default), all streams are sent on unique ports, with each stream on a port 2 numbers higher than the previous. VLC/Live555 requires this to be set to 1, to be able to receive the stream. The RTP stack in libavformat for receiving requires all streams to be sent on unique ports.
Example command lines follow.
To broadcast a stream on the local subnet, for watching in VLC:
ffmpeg -re -i input -f sap sap://224.0.0.255?same_port=1
Similarly, for watching in ffplay:
ffmpeg -re -i input -f sap sap://224.0.0.255
And for watching in ffplay, over IPv6:
ffmpeg -re -i input -f sap sap://[ff0e::1:2:3:4]
The syntax for a SAP url given to the demuxer is:
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]
Stream Control Transmission Protocol.
The accepted URL syntax is:
sctp://host:port[?options]
The protocol accepts the following options:
If set to any value, listen for an incoming connection. Outgoing connection is done by default.
Set the maximum number of streams. By default no limit is set.
Haivision Secure Reliable Transport Protocol via libsrt.
The supported syntax for a SRT URL is:
srt://hostname:port[?options]
options contains a list of &-separated options of the form key=val.
or
options srt://hostname:port
options contains a list of ’-key val’ options.
This protocol accepts the following options.
Connection timeout; SRT cannot connect for RTT > 1500 msec (2 handshake exchanges) with the default connect timeout of 3 seconds. This option applies to the caller and rendezvous connection modes. The connect timeout is 10 times the value set for the rendezvous mode (which can be used as a workaround for this connection problem with earlier versions).
Flight Flag Size (Window Size), in bytes. FFS is actually an internal parameter and you should set it to not less than ‘recv_buffer_size’ and ‘mss’. The default value is relatively large, therefore unless you set a very large receiver buffer, you do not need to change this option. Default value is 25600.
Sender nominal input rate, in bytes per seconds. Used along with ‘oheadbw’, when ‘maxbw’ is set to relative (0), to calculate maximum sending rate when recovery packets are sent along with the main media stream: ‘inputbw’ * (100 + ‘oheadbw’) / 100 if ‘inputbw’ is not set while ‘maxbw’ is set to relative (0), the actual input rate is evaluated inside the library. Default value is 0.
IP Type of Service. Applies to sender only. Default value is 0xB8.
IP Time To Live. Applies to sender only. Default value is 64.
Set socket listen timeout.
Maximum sending bandwidth, in bytes per seconds. -1 infinite (CSRTCC limit is 30mbps) 0 relative to input rate (see ‘inputbw’) >0 absolute limit value Default value is 0 (relative)
Connection mode. ‘caller’ opens client connection. ‘listener’ starts server to listen for incoming connections. ‘rendezvous’ use Rendez-Vous connection mode. Default value is caller.
Maximum Segment Size, in bytes. Used for buffer allocation and rate calculation using a packet counter assuming fully filled packets. The smallest MSS between the peers is used. This is 1500 by default in the overall internet. This is the maximum size of the UDP packet and can be only decreased, unless you have some unusual dedicated network settings. Default value is 1500.
If set to 1, Receiver will send ‘UMSG_LOSSREPORT‘ messages periodically until a lost packet is retransmitted or intentionally dropped. Default value is 1.
Recovery bandwidth overhead above input rate, in percents. See ‘inputbw’. Default value is 25%.
HaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. The passphrase is the shared secret between the sender and the receiver. It is used to generate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function). It is used only if ‘pbkeylen’ is non-zero. It is used on the receiver only if the received data is encrypted. The configured passphrase cannot be recovered (write-only).
Sender encryption key length, in bytes. Only can be set to 0, 16, 24 and 32. Enable sender encryption if not 0. Not required on receiver (set to 0), key size obtained from sender in HaiCrypt handshake. Default value is 0.
Set receive buffer size, expressed in bytes.
Set send buffer size, expressed in bytes.
Set raise error timeout for read/write optations.
This option is only relevant in read mode: if no data arrived in more than this time interval, raise error.
Too-late Packet Drop. When enabled on receiver, it skips missing packets that have not been delivered in time and delivers the following packets to the application when their time-to-play has come. It also sends a fake ACK to the sender. When enabled on sender and enabled on the receiving peer, the sender drops the older packets that have no chance of being delivered in time. It was automatically enabled in the sender if the receiver supports it.
Timestamp-based Packet Delivery Delay. Used to absorb burst of missed packet retransmission.
For more information see: https://github.com/Haivision/srt.
Secure Real-time Transport Protocol.
The accepted options are:
Select input and output encoding suites.
Supported values:
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.
Virtually extract a segment of a file or another stream. The underlying stream must be seekable.
Accepted options:
Start offset of the extracted segment, in bytes.
End offset of the extracted segment, in bytes. If set to 0, extract till end of file.
Examples:
Extract a chapter from a DVD VOB file (start and end sectors obtained externally and multiplied by 2048):
subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
Play an AVI file directly from a TAR archive:
subfile,,start,183241728,end,366490624,,:archive.tar
Play a MPEG-TS file from start offset till end:
subfile,,start,32815239,end,0,,:video.ts
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
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 for an incoming connection. Default value is 0.
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.
Set listen timeout, expressed in milliseconds.
Set receive buffer size, expressed bytes.
Set send buffer size, expressed bytes.
Set TCP_NODELAY to disable Nagle’s algorithm. Default value is 0.
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
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 AVOptions):
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.
If enabled, try to verify the peer that we are communicating with. Note, if using OpenSSL, this currently only makes sure that the peer certificate is signed by one of the root certificates in the CA database, but it does not validate that the certificate actually matches the host name we are trying to connect to. (With other backends, the host name is validated as well.)
This is disabled by default since it requires a CA database to be provided by the caller in many cases.
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.)
A file containing the private key for the certificate.
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
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.
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.
If set to nonzero, the output will have the specified constant bitrate if the input has enough packets to sustain it.
When using bitrate this specifies the maximum number of bits in packet bursts.
Override the local UDP port to bind with.
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.
Set the size in bytes of UDP packets.
Explicitly allow or disallow reusing UDP sockets.
Set the time to live value (for multicast only).
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.
Only receive packets sent to the multicast group from one of the specified sender IP addresses.
Ignore packets sent to the multicast group from the specified sender IP addresses.
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.
Survive in case of UDP receiving circular buffer overrun. Default value is 0.
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.
Explicitly allow or disallow UDP broadcasting.
Note that broadcasting may not work properly on networks having a broadcast storm protection.
ffmpeg to stream over UDP to a remote endpoint:
ffmpeg -i input -f format udp://hostname:port
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
ffmpeg to receive over UDP from a remote endpoint:
ffmpeg -i udp://[multicast-address]:port ...
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 AVOptions):
Timeout in ms.
Create the Unix socket in listening mode.
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 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 (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
Set the sample rate in Hz. Default is 48000.
Set the number of channels. Default is 2.
Android camera input device.
This input devices uses the Android Camera2 NDK API which is available on devices with API level 24+. The availability of android_camera is autodetected during configuration.
This device allows capturing from all cameras on an Android device, which are integrated into the Camera2 NDK API.
The available cameras are enumerated internally and can be selected with the camera_index parameter. The input file string is discarded.
Generally the back facing camera has index 0 while the front facing camera has index 1.
Set the video size given as a string such as 640x480 or hd720. Falls back to the first available configuration reported by Android if requested video size is not available or by default.
Set the video framerate. Falls back to the first available configuration reported by Android if requested framerate is not available or by default (-1).
Set the index of the camera to use. Default is 0.
Set the maximum number of frames to buffer. Default is 5.
AVFoundation input device.
AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.
The input filename has to be given in the following syntax:
-i "[[VIDEO]:[AUDIO]]"
The first entry selects the video input while the latter selects the audio input. The stream has to be specified by the device name or the device index as shown by the device list. Alternatively, the video and/or audio input device can be chosen by index using the ‘ -video_device_index <INDEX> ’ and/or ‘ -audio_device_index <INDEX> ’ , 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:
defaultSelect the AVFoundation default device of the corresponding type.
noneDo not record the corresponding media type. This is equivalent to specifying an empty device name or index.
AVFoundation supports the following options:
If set to true, a list of all available input devices is given showing all device names and indices.
Specify the video device by its index. Overrides anything given in the input filename.
Specify the audio device by its index. Overrides anything given in the input filename.
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
Set the grabbing frame rate. Default is ntsc, corresponding to a
frame rate of 30000/1001.
Set the video frame size.
Capture the mouse pointer. Default is 0.
Capture the screen mouse clicks. Default is 0.
$ ffmpeg -f avfoundation -list_devices true -i ""
$ ffmpeg -f avfoundation -i "0:0" out.avi
$ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
BSD video input device.
Set the frame rate.
Set the video frame size. Default is vga.
Available values are:
The decklink input device provides capture capabilities for Blackmagic DeckLink devices.
To enable this input device, you need the Blackmagic DeckLink SDK and you
need to configure with the appropriate --extra-cflags
and --extra-ldflags.
On Windows, you need to run the IDL files through widl.
DeckLink is very picky about the formats it supports. Pixel format of the
input can be set with ‘raw_format’.
Framerate and video size must be determined for your device with
-list_formats 1. Audio sample rate is always 48 kHz and the number
of channels can be 2, 8 or 16. Note that all audio channels are bundled in one single
audio track.
If set to ‘true’, print a list of devices and exit. Defaults to ‘false’.
If set to ‘true’, print a list of supported formats and exit. Defaults to ‘false’.
This sets the input video format to the format given by the FourCC. To see the supported values of your device(s) use ‘list_formats’. Note that there is a FourCC ‘'pal '’ that can also be used as ‘pal’ (3 letters). Default behavior is autodetection of the input video format, if the hardware supports it.
This is a deprecated option, you can use ‘raw_format’ instead. If set to ‘1’, video is captured in 10 bit v210 instead of uyvy422. Not all Blackmagic devices support this option.
Set the pixel format of the captured video. Available values are:
If set to nonzero, an additional teletext stream will be captured from the vertical ancillary data. Both SD PAL (576i) and HD (1080i or 1080p) sources are supported. In case of HD sources, OP47 packets are decoded.
This option is a bitmask of the SD PAL VBI lines captured, specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB in the mask. Selected lines which do not contain teletext information will be ignored. You can use the special ‘all’ constant to select all possible lines, or ‘standard’ to skip lines 6, 318 and 319, which are not compatible with all receivers.
For SD sources, ffmpeg needs to be compiled with --enable-libzvbi. For
HD sources, on older (pre-4K) DeckLink card models you have to capture in 10
bit mode.
Defines number of audio channels to capture. Must be ‘2’, ‘8’ or ‘16’. Defaults to ‘2’.
Sets the decklink device duplex mode. Must be ‘unset’, ‘half’ or ‘full’. Defaults to ‘unset’.
Sets the video input source. Must be ‘unset’, ‘sdi’, ‘hdmi’, ‘optical_sdi’, ‘component’, ‘composite’ or ‘s_video’. Defaults to ‘unset’.
Sets the audio input source. Must be ‘unset’, ‘embedded’, ‘aes_ebu’, ‘analog’, ‘analog_xlr’, ‘analog_rca’ or ‘microphone’. Defaults to ‘unset’.
Sets the video packet timestamp source. Must be ‘video’, ‘audio’, ‘reference’, ‘wallclock’ or ‘abs_wallclock’. Defaults to ‘video’.
Sets the audio packet timestamp source. Must be ‘video’, ‘audio’, ‘reference’, ‘wallclock’ or ‘abs_wallclock’. Defaults to ‘audio’.
If set to ‘true’, color bars are drawn in the event of a signal loss. Defaults to ‘true’.
Sets maximum input buffer size in bytes. If the buffering reaches this value, incoming frames will be dropped. Defaults to ‘1073741824’.
Sets the audio sample bit depth. Must be ‘16’ or ‘32’. Defaults to ‘16’.
If set to ‘true’, timestamps are forwarded as they are without removing the initial offset. Defaults to ‘false’.
ffmpeg -f decklink -list_devices 1 -i dummy
ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi
ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
KMS video input device.
Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a DRM object that can be passed to other hardware functions.
Requires either DRM master or CAP_SYS_ADMIN to run.
If you don’t understand what all of that means, you probably don’t want this. Look at ‘x11grab’ instead.
DRM device to capture on. Defaults to ‘/dev/dri/card0’.
Pixel format of the framebuffer. Defaults to ‘bgr0’.
Format modifier to signal on output frames. This is necessary to import correctly into some APIs, but can’t be autodetected. See the libdrm documentation for possible values.
KMS CRTC ID to define the capture source. The first active plane on the given CRTC will be used.
KMS plane ID to define the capture source. Defaults to the first active plane found if neither ‘crtc_id’ nor ‘plane_id’ are specified.
Framerate to capture at. This is not synchronised to any page flipping or framebuffer
changes - it just defines the interval at which the framebuffer is sampled. Sampling
faster than the framebuffer update rate will generate independent frames with the same
content. Defaults to 30.
ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
The libndi_newtek input device provides capture capabilities for using NDI (Network Device Interface, standard created by NewTek).
Input filename is a NDI source name that could be found by sending -find_sources 1 to command line - it has no specific syntax but human-readable formatted.
To enable this input device, you need the NDI SDK and you
need to configure with the appropriate --extra-cflags
and --extra-ldflags.
If set to ‘true’, print a list of found/available NDI sources and exit. Defaults to ‘false’.
Override time to wait until the number of online sources have changed. Defaults to ‘0.5’.
When this flag is ‘false’, all video that you receive will be progressive. Defaults to ‘true’.
ffmpeg -f libndi_newtek -find_sources 1 -i dummy
ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
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..
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.
Set the video size in the captured video.
Set the frame rate in the captured video.
Set the sample rate (in Hz) of the captured audio.
Set the sample size (in bits) of the captured audio.
Set the number of channels in the captured audio.
If set to ‘true’, print a list of devices and exit.
If set to ‘true’, print a list of selected device’s options and exit.
Set video device number for devices with the same name (starts at 0, defaults to 0).
Set audio device number for devices with the same name (starts at 0, defaults to 0).
Select pixel format to be used by DirectShow. This may only be set when the video codec is not set or set to rawvideo.
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
Select video capture pin to use by name or alternative name.
Select audio capture pin to use by name or alternative name.
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.
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.
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.
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.
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.
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.
If set to ‘true’, before capture starts, popup a display dialog to the end user, allowing them to manually modify TV channels and frequencies.
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).
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.
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.
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.
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.
$ ffmpeg -list_devices true -f dshow -i dummy
$ ffmpeg -f dshow -i video="Camera"
$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
$ ffmpeg -f dshow -i video="Camera":audio="Microphone"
$ ffmpeg -list_options true -f dshow -i video="Camera"
$ 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"
$ 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"
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
Set the frame rate. Default is 25.
Win32 GDI-based screen capture device.
This device allows you to capture a region of the display on Windows.
There are two options for the input filename:
desktop
or
title=window_title
The first option will capture the entire desktop, or a fixed region of the desktop. The second option will instead capture the contents of a single window, regardless of its position on the screen.
For example, to grab the entire desktop using ffmpeg:
ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg
Grab a 640x480 region at position 10,20:
ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg
Grab the contents of the window named "Calculator"
ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg
Specify whether to draw the mouse pointer. Use the value 0 to
not draw the pointer. Default value is 1.
Set the grabbing frame rate. Default value is ntsc,
corresponding to a frame rate of 30000/1001.
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
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.
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.
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.
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.
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.
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.
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.
ffplay -f iec61883 -i auto
ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
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/
Set the number of channels. Default is 2.
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’.
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.
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.
Dump graph to stderr.
ffplay:
ffplay -f lavfi -graph "color=c=pink [out0]" dummy
ffplay -f lavfi color=c=pink
ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
ffplay:
ffplay -f lavfi "amovie=test.wav"
ffplay:
ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin
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
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.
Set paranoia recovery mode flags. It accepts one of the following values:
Default value is ‘disable’.
For more information about the available recovery modes, consult the paranoia project documentation.
IIDC1394 input device, based on libdc1394 and libraw1394.
Requires the configure option --enable-libdc1394.
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:
The official Windows implementation, providing hardware acceleration with supported devices and software fallback. See http://openal.org/.
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.
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.
Set the number of channels in the captured audio. Only the values ‘1’ (monaural) and ‘2’ (stereo) are currently supported. Defaults to ‘2’.
Set the sample size (in bits) of the captured audio. Only the values ‘8’ and ‘16’ are currently supported. Defaults to ‘16’.
Set the sample rate (in Hz) of the captured audio. Defaults to ‘44.1k’.
If set to ‘true’, print a list of devices and exit. Defaults to ‘false’.
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.
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
Set the sample rate in Hz. Default is 48000.
Set the number of channels. Default is 2.
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.
Connect to a specific PulseAudio server, specified by an IP address. Default server is used when not provided.
Specify the application name PulseAudio will use when showing active clients,
by default it is the LIBAVFORMAT_IDENT string.
Specify the stream name PulseAudio will use when showing active streams, by default it is "record".
Specify the samplerate in Hz, by default 48kHz is used.
Specify the channels in use, by default 2 (stereo) is set.
Specify the number of bytes per frame, by default it is set to 1024.
Specify the minimal buffering fragment in PulseAudio, it will affect the audio latency. By default it is unset.
Set the initial PTS using the current time. Default is 1.
Record a stream from default device:
ffmpeg -f pulse -i default /tmp/pulse.wav
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
Set the sample rate in Hz. Default is 48000.
Set the number of channels. Default is 2.
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:
ffplay -f video4linux2 -list_formats all /dev/video0
ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
For more information about Video4Linux, check http://linuxtv.org/.
Set the standard. Must be the name of a supported standard. To get a list of the supported standards, use the ‘list_standards’ option.
Set the input channel number. Default to -1, which means using the previously selected channel.
Set the video frame size. The argument must be a string in the form WIDTHxHEIGHT or a valid size abbreviation.
Select the pixel format (only valid for raw video input).
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.
Set the preferred video frame rate.
List available formats (supported pixel formats, codecs, and frame sizes) and exit.
Available values are:
Show all available (compressed and non-compressed) formats.
Show only raw video (non-compressed) formats.
Show only compressed formats.
List supported standards and exit.
Available values are:
Show all supported standards.
Set type of timestamps for grabbed frames.
Available values are:
Use timestamps from the kernel.
Use absolute timestamps (wall clock).
Force conversion from monotonic to absolute timestamps.
Default value is default.
Use libv4l2 (v4l-utils) conversion functions. Default is 0.
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.
Set the video frame size.
Set the grabbing frame rate. Default value is ntsc,
corresponding to a frame rate of 30000/1001.
X11 video input device.
To enable this input device during configuration you need libxcb installed on your system. It will be automatically detected during configuration.
This device allows one to capture a region of an X11 display.
The filename passed as input has the syntax:
[hostname]:display_number.screen_number[+x_offset,y_offset]
hostname:display_number.screen_number specifies the
X11 display name of the screen to grab from. hostname can be
omitted, and defaults to "localhost". The environment variable
DISPLAY contains the default display name.
x_offset and y_offset specify the offsets of the grabbed area with respect to the top-left border of the X11 screen. They default to 0.
Check the X11 documentation (e.g. man X) for more detailed
information.
Use the xdpyinfo program for getting basic information about
the properties of your X11 display (e.g. grep for "name" or
"dimensions").
For example to grab from ‘:0.0’ using ffmpeg:
ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg
Grab at position 10,20:
ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
Specify whether to draw the mouse pointer. A value of 0 specifies
not to draw the pointer. Default value is 1.
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
Set the grabbing frame rate. Default value is ntsc,
corresponding to a frame rate of 30000/1001.
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.
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
Set the video frame size. Default value is vga.
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 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 (Advanced Linux Sound Architecture) output device.
ffmpeg -i INPUT -f alsa default
ffmpeg -i INPUT -f alsa hw:1,7
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
Set the CACA window title, if not specified default to the filename specified for the output device.
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.
Set display driver.
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.
Set antialias method. Antialiasing smoothens the rendered
image and avoids the commonly seen staircase effect.
The accepted values are listed with -list_dither antialiases.
Set which characters are going to be used when rendering text.
The accepted values are listed with -list_dither charsets.
Set color to be used when rendering text.
The accepted values are listed with -list_dither colors.
If set to ‘true’, print a list of available drivers and exit.
List available dither options related to the argument.
The argument must be one of algorithms, antialiases,
charsets, colors.
ffmpeg output is an
CACA window, forcing its size to 80x25:
ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
The decklink output device provides playback capabilities for Blackmagic DeckLink devices.
To enable this output device, you need the Blackmagic DeckLink SDK and you
need to configure with the appropriate --extra-cflags
and --extra-ldflags.
On Windows, you need to run the IDL files through widl.
DeckLink is very picky about the formats it supports. Pixel format is always
uyvy422, framerate, field order and video size must be determined for your
device with -list_formats 1. Audio sample rate is always 48 kHz.
If set to ‘true’, print a list of devices and exit. Defaults to ‘false’.
If set to ‘true’, print a list of supported formats and exit. Defaults to ‘false’.
Amount of time to preroll video in seconds. Defaults to ‘0.5’.
ffmpeg -i test.avi -f decklink -list_devices 1 dummy
ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
The libndi_newtek output device provides playback capabilities for using NDI (Network Device Interface, standard created by NewTek).
Output filename is a NDI name.
To enable this output device, you need the NDI SDK and you
need to configure with the appropriate --extra-cflags
and --extra-ldflags.
NDI uses uyvy422 pixel format natively, but also supports bgra, bgr0, rgba and rgb0.
The audio reference level in dB. This specifies how many dB above the reference level (+4dBU) is the full range of 16 bit audio. Defaults to ‘0’.
These specify whether video "clock" themselves. Defaults to ‘false’.
These specify whether audio "clock" themselves. Defaults to ‘false’.
ffmpeg -i "udp://@239.1.1.1:10480?fifo_size=1000000&overrun_nonfatal=1" -vf "scale=720:576,fps=fps=25,setdar=dar=16/9,format=pix_fmts=uyvy422" -f libndi_newtek NEW_NDI1
Linux framebuffer output device.
The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on a computer monitor, typically on the console. It is accessed through a file device node, usually ‘/dev/fb0’.
For more detailed information read the file ‘Documentation/fb/framebuffer.txt’ included in the Linux source tree.
Set x/y coordinate of top left corner. Default is 0.
Play a file on framebuffer device ‘/dev/fb0’. Required pixel format depends on current framebuffer settings.
ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0
See also http://linux-fbdev.sourceforge.net/, and fbset(1).
OpenGL 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.
Set background color. Black is a default.
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.
Set the SDL window title, if not specified default to the filename specified for the output device. Ignored when ‘no_window’ is set.
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.
Play a file on SDL window using OpenGL rendering:
ffmpeg -i INPUT -f opengl "window title"
OSS (Open Sound System) output device.
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
Connect to a specific PulseAudio server, specified by an IP address. Default server is used when not provided.
Specify the application name PulseAudio will use when showing active clients,
by default it is the LIBAVFORMAT_IDENT string.
Specify the stream name PulseAudio will use when showing active streams, by default it is set to the specified output name.
Specify the device to use. Default device is used when not provided.
List of output devices can be obtained with command pactl list sinks.
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.
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).
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.
Play a file on default device on default server:
ffmpeg -i INPUT -f pulse "stream name"
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/
Set the SDL window title, if not specified default to the filename specified for the output device.
Set the name of the iconified SDL window, if not specified it is set to the same value of window_title.
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.
Set fullscreen mode when non-zero value is provided. Default value is zero.
The window created by the device can be controlled through the following interactive commands.
Quit the device immediately.
The following command shows the ffmpeg output is an
SDL window, forcing its size to the qcif format:
ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
sndio audio output device.
XV (XVideo) output device.
This output device allows one to show a video stream in a X Window System window.
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.
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.
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.
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.
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/.
ffmpeg at the
same time:
ffmpeg -i INPUT OUTPUT -f xv display
ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
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.
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.
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.
Set the number of used input channels. Default value is 0. This option is only used for special remapping.
Set the input sample rate. Default value is 0.
Set the output sample rate. Default value is 0.
Specify the input sample format. It is set by default to none.
Specify the output sample format. It is set by default to none.
Set the internal sample format. Default value is none.
This will automatically be chosen when it is not explicitly set.
Set the input/output channel layout.
See (ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.
Set the center mix level. It is a value expressed in deciBel, and must be in the interval [-32,32].
Set the surround mix level. It is a value expressed in deciBel, and must be in the interval [-32,32].
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].
Set rematrix volume. Default value is 1.0.
Set maximum output value for rematrixing. This can be used to prevent clipping vs. preventing volume reduction. A value of 1.0 prevents clipping.
Set flags used by the converter. Default value is 0.
It supports the following individual flags:
force resampling, this flag forces resampling to be used even when the input and output sample rates match.
Set the dither scale. Default value is 1.
Set dither method. Default value is 0.
Supported values:
select rectangular dither
select triangular dither
select triangular dither with high pass
select Lipshitz noise shaping dither.
select Shibata noise shaping dither.
select low Shibata noise shaping dither.
select high Shibata noise shaping dither.
select f-weighted noise shaping dither
select modified-e-weighted noise shaping dither
select improved-e-weighted noise shaping dither
Set resampling engine. Default value is swr.
Supported values:
select the native SW Resampler; filter options precision and cheby are not applicable in this case.
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.
For swr only, set resampling filter size, default value is 32.
For swr only, set resampling phase shift, default value is 10, and must be in the interval [0,30].
Use linear interpolation when enabled (the default). Disable it if you want to preserve speed instead of quality when exact_rational fails.
For swr only, when enabled, try to use exact phase_count based on input and
output sample rate. However, if it is larger than 1 << phase_shift,
the phase_count will be 1 << phase_shift as fallback. Default is enabled.
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).
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’.
For soxr only, selects passband rolloff none (Chebyshev) & higher-precision approximation for ’irrational’ ratios. Default value is 0.
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.
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.
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).
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.
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.
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.
Select matrixed stereo encoding.
It accepts the following values:
select none
select Dolby
select Dolby Pro Logic II
Default value is none.
For swr only, select resampling filter type. This only affects resampling operations.
It accepts the following values:
select cubic
select Blackman Nuttall windowed sinc
select Kaiser windowed sinc
For swr only, set Kaiser window beta value. Must be a double float value in the interval [2,16], default value is 9.
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.
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.
Set the scaler flags. This is also used to set the scaling algorithm. Only a single algorithm should be selected. Default value is ‘bicubic’.
It accepts the following values:
Select fast bilinear scaling algorithm.
Select bilinear scaling algorithm.
Select bicubic scaling algorithm.
Select experimental scaling algorithm.
Select nearest neighbor rescaling algorithm.
Select averaging area rescaling algorithm.
Select bicubic scaling algorithm for the luma component, bilinear for chroma components.
Select Gaussian rescaling algorithm.
Select sinc rescaling algorithm.
Select Lanczos rescaling algorithm.
Select natural bicubic spline rescaling algorithm.
Enable printing/debug logging.
Enable accurate rounding.
Enable full chroma interpolation.
Select full chroma input.
Enable bitexact output.
Set source width.
Set source height.
Set destination width.
Set destination height.
Set source pixel format (must be expressed as an integer).
Set destination pixel format (must be expressed as an integer).
Select source range.
Select destination range.
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.
Set the dithering algorithm. Accepts one of the following values. Default value is ‘auto’.
automatic choice
no dithering
bayer dither
error diffusion dither
arithmetic dither, based using addition
arithmetic dither, based using xor (more random/less apparent patterning that a_dither).
Set the alpha blending to use when the input has alpha but the output does not. Default value is ‘none’.
Blend onto a uniform background color
Blend onto a checkerboard
No blending
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.
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.
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".
A filtergraph has a textual representation, which is recognized by the
‘-filter’/‘-vf’/‘-af’ and
‘-filter_complex’ options in ffmpeg and
‘-vf’/‘-af’ in ffplay, and by the
avfilter_graph_parse_ptr() function defined in
‘libavfilter/avfilter.h’.
A filterchain consists of a sequence of connected filters, each one connected to the previous one in the sequence. A filterchain is represented by a list of ","-separated filter descriptions.
A filtergraph consists of a sequence of filterchains. A sequence of filterchains is represented by a list of ";"-separated filterchain descriptions.
A filter is represented by a string of the form: [in_link_1]...[in_link_N]filter_name@id=arguments[out_link_1]...[out_link_M]
filter_name is the name of the filter class of which the described filter is an instance of, and has to be the name of one of the filter classes registered in the program optionally followed by "@id". The name of the filter class is optionally followed by a string "=arguments".
arguments is a string which contains the parameters used to initialize the filter instance. It may have one of two forms:
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’.
If the option value itself is a list of items (e.g. the format filter
takes a list of pixel formats), the items in the list are usually separated by
‘|’.
The list of arguments can be quoted using the character ‘'’ as initial and ending mark, and the character ‘\’ for escaping the characters within the quoted text; otherwise the argument string is considered terminated when the next special character (belonging to the set ‘[]=;,’) is encountered.
The name and arguments of the filter are optionally preceded and followed by a list of link labels. A link label allows one to name a link and associate it to a filter output or input pad. The preceding labels in_link_1 ... in_link_N, are associated to the filter input pads, the following labels out_link_1 ... out_link_M, are associated to the output pads.
When two link labels with the same name are found in the filtergraph, a link between the corresponding input and output pad is created.
If an output pad is not labelled, it is linked by default to the first unlabelled input pad of the next filter in the filterchain. For example in the filterchain
nullsrc, split[L1], [L2]overlay, nullsink
the split filter instance has two output pads, and the overlay filter instance two input pads. The first output pad of split is labelled "L1", the first input pad of overlay is labelled "L2", and the second output pad of split is linked to the second input pad of overlay, which are both unlabelled.
In a filter description, if the input label of the first filter is not specified, "in" is assumed; if the output label of the last filter is not specified, "out" is assumed.
In a complete filterchain all the unlabelled filter input and output pads must be connected. A filtergraph is considered valid if all the filter input and output pads of all the filterchains are connected.
Libavfilter will automatically insert scale filters where format
conversion is required. It is possible to specify swscale flags
for those automatically inserted scalers by prepending
sws_flags=flags;
to the filtergraph description.
Here is a BNF description of the filtergraph syntax:
NAME ::= sequence of alphanumeric characters and '_' FILTER_NAME ::= NAME["@"NAME] LINKLABEL ::= "[" NAME "]" LINKLABELS ::= LINKLABEL [LINKLABELS] FILTER_ARGUMENTS ::= sequence of chars (possibly quoted) FILTER ::= [LINKLABELS] FILTER_NAME ["=" FILTER_ARGUMENTS] [LINKLABELS] FILTERCHAIN ::= FILTER [,FILTERCHAIN] FILTERGRAPH ::= [sws_flags=flags;] FILTERCHAIN [;FILTERGRAPH]
Filtergraph description composition entails several levels of escaping. See (ffmpeg-utils)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"
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:
timestamp expressed in seconds, NAN if the input timestamp is unknown
sequential number of the input frame, starting from 0
the position in the file of the input frame, NAN if unknown
width and height of the input frame if video
Additionally, these filters support an ‘enable’ command that can be used to re-define the expression.
Like any other filtering option, the ‘enable’ option follows the same rules.
For example, to enable a blur filter (smartblur) from 10 seconds to 3 minutes, and a curves filter starting at 3 seconds:
smartblur = enable='between(t,10,3*60)', curves = enable='gte(t,3)' : preset=cross_process
See ffmpeg -filters to view which filters have timeline support.
Some filters with several inputs support a common set of options. These options can only be set by name, not with the short notation.
The action to take when EOF is encountered on the secondary input; it accepts one of the following values:
Repeat the last frame (the default).
End both streams.
Pass the main input through.
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.
If set to 1, force the filter to extend the last frame of secondary streams until the end of the primary stream. A value of 0 disables this behavior. Default value is 1.
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.
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:
Set input gain. Default is 1. Range is between 0.015625 and 64.
If a signal of stream rises above this level it will affect the gain reduction. By default it is 0.125. Range is between 0.00097563 and 1.
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.
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.
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.
Set the amount by how much signal will be amplified after processing. Default is 1. Range is from 1 to 64.
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.
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.
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.
How much to use compressed signal in output. Default is 1. Range is between 0 and 1.
Simple audio dynamic range commpression/expansion filter.
The filter accepts the following options:
Set contrast. Default is 33. Allowed range is between 0 and 100.
Copy the input audio source unchanged to the output. This is mainly useful for testing purposes.
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:
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.
Specify the duration of the cross fade effect. See (ffmpeg-utils)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.
Should first stream end overlap with second stream start. Default is enabled.
Set curve for cross fade transition for first stream.
Set curve for cross fade transition for second stream.
For description of available curve types see afade filter description.
ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac
ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac
Reduce audio bit resolution.
This filter is bit crusher with enhanced functionality. A bit crusher is used to audibly reduce number of bits an audio signal is sampled with. This doesn’t change the bit depth at all, it just produces the effect. Material reduced in bit depth sounds more harsh and "digital". This filter is able to even round to continuous values instead of discrete bit depths. Additionally it has a D/C offset which results in different crushing of the lower and the upper half of the signal. An Anti-Aliasing setting is able to produce "softer" crushing sounds.
Another feature of this filter is the logarithmic mode. This setting switches from linear distances between bits to logarithmic ones. The result is a much more "natural" sounding crusher which doesn’t gate low signals for example. The human ear has a logarithmic perception, so this kind of crushing is much more pleasant. Logarithmic crushing is also able to get anti-aliased.
The filter accepts the following options:
Set level in.
Set level out.
Set bit reduction.
Set mixing amount.
Can be linear: lin or logarithmic: log.
Set DC.
Set anti-aliasing.
Set sample reduction.
Enable LFO. By default disabled.
Set LFO range.
Set LFO rate.
Delay one or more audio channels.
Samples in delayed channel are filled with silence.
The filter accepts the following option:
Set list of delays in milliseconds for each channel separated by ’|’. Unused delays will be silently ignored. If number of given delays is smaller than number of channels all remaining channels will not be delayed. If you want to delay exact number of samples, append ’S’ to number.
adelay=1500|0|500
adelay=0|500S|700S
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.
Set input gain of reflected signal. Default is 0.6.
Set output gain of reflected signal. Default is 0.3.
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.
Set list of loudness of reflected signals separated by ’|’.
Allowed range for each decay is (0 - 1.0].
Default is 0.5.
aecho=0.8:0.88:60:0.4
aecho=0.8:0.88:6:0.4
aecho=0.8:0.9:1000:0.3
aecho=0.8:0.9:1000|1800:0.3|0.25
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:
Set input gain.
Set output gain.
Set filter mode. For restoring material use reproduction mode, otherwise
use production mode. Default is reproduction mode.
Set filter type. Selects medium. Can be one of the following:
select Columbia.
select EMI.
select BSI (78RPM).
select RIAA.
select Compact Disc (CD).
select 50µs (FM).
select 75µs (FM).
select 50µs (FM-KF).
select 75µs (FM-KF).
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:
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.
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:
channel number of the current expression
number of the evaluated sample, starting from 0
sample rate
time of the evaluated sample expressed in seconds
input and output number of channels
the value of input channel with number CH
Note: this filter is slow. For faster processing you should use a dedicated filter.
aeval=val(ch)/2:c=same
aeval=val(0)|-val(1)
Apply fade-in/out effect to input audio.
A description of the accepted parameters follows.
Specify the effect type, can be either in for fade-in, or
out for a fade-out effect. Default is in.
Specify the number of the start sample for starting to apply the fade effect. Default is 0.
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.
Specify the start time of the fade effect. Default is 0. The value must be specified as a time duration; see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. If set this option is used instead of start_sample.
Specify the duration of the fade effect. See (ffmpeg-utils)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.
Set curve for fade transition.
It accepts the following values:
select triangular, linear slope (default)
select quarter of sine wave
select half of sine wave
select exponential sine wave
select logarithmic
select inverted parabola
select quadratic
select cubic
select square root
select cubic root
select parabola
select exponential
select inverted quarter of sine wave
select inverted half of sine wave
select double-exponential seat
select double-exponential sigmoid
afade=t=in:ss=0:d=15
afade=t=out:st=875:d=25
Apply arbitrary expressions to samples in frequency domain.
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.
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:
sample rate
current frequency bin number
number of available bins
channel number of the current expression
number of channels
current frame pts
Set window size.
It accepts the following values:
Default is w4096
Set window function. Default is hann.
Set window overlap. If set to 1, the recommended overlap for selected
window function will be picked. Default is 0.75.
afftfilt="1-clip((b/nb)*b,0,1)"
Apply an arbitrary Frequency Impulse Response filter.
This filter is designed for applying long FIR filters, up to 30 seconds long.
It can be used as component for digital crossover filters, room equalization, cross talk cancellation, wavefield synthesis, auralization, ambiophonics and ambisonics.
This filter uses second stream as FIR coefficients. If second stream holds single channel, it will be used for all input channels in first stream, otherwise number of channels in second stream must be same as number of channels in first stream.
It accepts the following parameters:
Set dry gain. This sets input gain.
Set wet gain. This sets final output gain.
Set Impulse Response filter length. Default is 1, which means whole IR is processed.
Enable applying gain measured from power of IR.
ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
Set output format constraints for the input audio. The framework will negotiate the most appropriate format to minimize conversions.
It accepts the following parameters:
A ’|’-separated list of requested sample formats.
A ’|’-separated list of requested sample rates.
A ’|’-separated list of requested channel layouts.
See (ffmpeg-utils)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
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.
Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1.
If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.
Set a ratio by which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
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.
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.
Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.
Choose if exact signal should be taken for detection or an RMS like one.
Default is rms. Can be peak or rms.
Choose if the average level between all channels or the louder channel affects
the reduction.
Default is average. Can be average or maximum.
Apply an arbitrary Infinite Impulse Response filter.
It accepts the following parameters:
Set numerator/zeros coefficients.
Set denominator/poles coefficients.
Set channels gains.
Set input gain.
Set output gain.
Set coefficients format.
transfer function
Z-plane zeros/poles, cartesian (default)
Z-plane zeros/poles, polar radians
Z-plane zeros/poles, polar degrees
Set kind of processing.
Can be d - direct or s - serial cascading. Defauls is s.
Set filtering precision.
double-precision floating-point (default)
single-precision floating-point
32-bit integers
16-bit integers
Coefficients in tf format are separated by spaces and are in ascending
order.
Coefficients in zp format are separated by spaces and order of coefficients
doesn’t matter. Coefficients in zp format are complex numbers with i
imaginary unit.
Different coefficients and gains can be provided for every channel, in such case use ’|’ to separate coefficients or gains. Last provided coefficients will be used for all remaining channels.
aiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf:r=d
zp format:
aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s
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:
Set input gain. Default is 1.
Set output gain. Default is 1.
Don’t let signals above this level pass the limiter. Default is 1.
The limiter will reach its attenuation level in this amount of time in milliseconds. Default is 5 milliseconds.
Come back from limiting to attenuation 1.0 in this amount of milliseconds. Default is 50 milliseconds.
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.
Select how much the release time is affected by ASC, 0 means nearly no changes in release time while 1 produces higher release times.
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.
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:
Set frequency in Hz.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Specify the band-width of a filter in width_type units.
Specify which channels to filter, by default all available are filtered.
This filter supports the following commands:
Change allpass frequency. Syntax for the command is : "frequency"
Change allpass width_type. Syntax for the command is : "width_type"
Change allpass width. Syntax for the command is : "width"
Loop audio samples.
The filter accepts the following options:
Set the number of loops. Setting this value to -1 will result in infinite loops. Default is 0.
Set maximal number of samples. Default is 0.
Set first sample of loop. Default is 0.
Merge two or more audio streams into a single multi-channel stream.
The filter accepts the following options:
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.
amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
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
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:
The number of inputs. If unspecified, it defaults to 2.
How to determine the end-of-stream.
The duration of the longest input. (default)
The duration of the shortest input.
The duration of the first input.
The transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.
Specify weight of each input audio stream as sequence. Each weight is separated by space. By default all inputs have same weight.
High-order parametric multiband equalizer for each channel.
It accepts the following parameters:
This option string is in format: "cchn f=cf w=w g=g t=f | ..." Each equalizer band is separated by ’|’.
Set channel number to which equalization will be applied. If input doesn’t have that channel the entry is ignored.
Set central frequency for band. If input doesn’t have that frequency the entry is ignored.
Set band width in hertz.
Set band gain in dB.
Set filter type for band, optional, can be:
Butterworth, this is default.
Chebyshev type 1.
Chebyshev type 2.
With this option activated frequency response of anequalizer is displayed in video stream.
Set video stream size. Only useful if curves option is activated.
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.
Set frequency scale used to draw frequency response in video output. Can be linear or logarithmic. Default is logarithmic.
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.
anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
This filter supports the following commands:
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=...
Pass the audio source unchanged to the output.
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.
Set silence packet size. Default value is 4096.
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’.
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.
apad=pad_len=1024
apad=whole_len=10000
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
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.
Set input gain. Default is 0.4.
Set output gain. Default is 0.74
Set delay in milliseconds. Default is 3.0.
Set decay. Default is 0.4.
Set modulation speed in Hz. Default is 0.5.
Set modulation type. Default is triangular.
It accepts the following values:
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:
Set input gain. By default it is 1. Range is [0.015625 - 64].
Set output gain. By default it is 1. Range is [0.015625 - 64].
Set waveform shape the LFO will use. Can be one of: sine, triangle, square, sawup or sawdown. Default is sine.
Set modulation. Define how much of original signal is affected by the LFO.
Set left channel offset. Default is 0. Allowed range is [0 - 1].
Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
Set pulse width. Default is 1. Allowed range is [0 - 2].
Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing is set to bpm.
Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing is set to ms.
Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used if timing is set to hz.
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)"Resampler Options" section in the ffmpeg-resampler(1) manual for the complete list of supported options.
aresample=44100
aresample=async=1000
Reverse an audio clip.
Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.
atrim=end=5,areverse
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:
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.
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
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:
Set the output sample rate. Default is 44100 Hz.
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:
The (sequential) number of the input frame, starting from 0.
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.
The presentation timestamp of the input frame in seconds.
position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic audio)
The sample format.
The channel layout.
The sample rate for the audio frame.
The number of samples (per channel) in the frame.
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.
A list of Adler-32 checksums for each data plane.
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:
Short window length in seconds, used for peak and trough RMS measurement.
Default is 0.05 (50 milliseconds). Allowed range is [0.01 - 10].
Set metadata injection. All the metadata keys are prefixed with lavfi.astats.X,
where X is channel number starting from 1 or string Overall. Default is
disabled.
Available keys for each channel are: DC_offset Min_level Max_level Min_difference Max_difference Mean_difference RMS_difference Peak_level RMS_peak RMS_trough Crest_factor Flat_factor Peak_count Bit_depth Dynamic_range
and for Overall: DC_offset Min_level Max_level Min_difference Max_difference Mean_difference RMS_difference Peak_level RMS_level RMS_peak RMS_trough Flat_factor Peak_count Bit_depth Number_of_samples
For example full key look like this lavfi.astats.1.DC_offset or
this lavfi.astats.Overall.Peak_count.
For description what each key means read below.
Set number of frame after which stats are going to be recalculated. Default is disabled.
A description of each shown parameter follows:
Mean amplitude displacement from zero.
Minimal sample level.
Maximal sample level.
Minimal difference between two consecutive samples.
Maximal difference between two consecutive samples.
Mean difference between two consecutive samples. The average of each difference between two consecutive samples.
Root Mean Square difference between two consecutive samples.
Standard peak and RMS level measured in dBFS.
Peak and trough values for RMS level measured over a short window.
Standard ratio of peak to RMS level (note: not in dB).
Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels (i.e. either Min level or Max level).
Number of occasions (not the number of samples) that the signal attained either Min level or Max level.
Overall bit depth of audio. Number of bits used for each sample.
Measured dynamic range of audio in dB.
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.
atempo=0.8
atempo=1.25
Trim the input so that the output contains one continuous subpart of the input.
It accepts the following parameters:
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.
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.
Same as start, except this option sets the start timestamp in samples instead of seconds.
Same as end, except this option sets the end timestamp in samples instead of seconds.
The maximum duration of the output in seconds.
The number of the first sample that should be output.
The number of the first sample that should be dropped.
‘start’, ‘end’, and ‘duration’ are expressed as time duration specifications; see (ffmpeg-utils)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:
ffmpeg -i INPUT -af atrim=60:120
ffmpeg -i INPUT -af atrim=end_sample=1000
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:
Set the filter’s central frequency. Default is 3000.
Constant skirt gain if set to 1. Defaults to 0.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Specify the band-width of a filter in width_type units.
Specify which channels to filter, by default all available are filtered.
This filter supports the following commands:
Change bandpass frequency. Syntax for the command is : "frequency"
Change bandpass width_type. Syntax for the command is : "width_type"
Change bandpass width. Syntax for the command is : "width"
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:
Set the filter’s central frequency. Default is 3000.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Specify the band-width of a filter in width_type units.
Specify which channels to filter, by default all available are filtered.
This filter supports the following commands:
Change bandreject frequency. Syntax for the command is : "frequency"
Change bandreject width_type. Syntax for the command is : "width_type"
Change bandreject width. Syntax for the command is : "width"
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:
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.
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.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Determine how steep is the filter’s shelf transition.
Specify which channels to filter, by default all available are filtered.
This filter supports the following commands:
Change bass frequency. Syntax for the command is : "frequency"
Change bass width_type. Syntax for the command is : "width_type"
Change bass width. Syntax for the command is : "width"
Change bass gain. Syntax for the command is : "gain"
Apply a biquad IIR filter with the given coefficients. Where b0, b1, b2 and a0, a1, a2 are the numerator and denominator coefficients respectively. and channels, c specify which channels to filter, by default all available are filtered.
This filter supports the following commands:
Change biquad parameter. Syntax for the command is : "value"
Bauer stereo to binaural transformation, which improves headphone listening of stereo audio records.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libbs2b.
It accepts the following parameters:
Pre-defined crossfeed level.
Default level (fcut=700, feed=50).
Chu Moy circuit (fcut=700, feed=60).
Jan Meier circuit (fcut=650, feed=95).
Cut frequency (in Hz).
Feed level (in Hz).
Remap input channels to new locations.
It accepts the following parameters:
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.
The channel layout of the output stream.
If no mapping is present, the filter will implicitly map input channels to output channels, preserving indices.
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.
ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav
Split each channel from an input audio stream into a separate output stream.
It accepts the following parameters:
The channel layout of the input stream. The default is "stereo".
A channel layout describing the channels to be extracted as separate output streams or "all" to extract each input channel as a separate stream. The default is "all".
Choosing channels not present in channel layout in the input will result in an error.
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.
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
ffmpeg -i in.wav -filter_complex 'channelsplit=channel_layout=5.1:channels=LFE[LFE]' -map '[LFE]' lfe.wav
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:
Set input gain. Default is 0.4.
Set output gain. Default is 0.4.
Set delays. A typical delay is around 40ms to 60ms.
Set decays.
Set speeds.
Set depths.
chorus=0.7:0.9:55:0.4:0.25:2
chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
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
Compress or expand the audio’s dynamic range.
It accepts the following parameters:
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.
A list of points for the transfer function, specified in dB relative to the
maximum possible signal amplitude. Each key points list must be defined using
the following syntax: x0/y0|x1/y1|x2/y2|.... or
x0/y0 x1/y1 x2/y2 ....
The input values must be in strictly increasing order but the transfer function
does not have to be monotonically rising. The point 0/0 is assumed but
may be overridden (by 0/out-dBn). Typical values for the transfer
function are -70/-70|-60/-20|1/0.
Set the curve radius in dB for all joints. It defaults to 0.01.
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.
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.
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.
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
compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
compand=points=-80/-80|-6/-6|0/-3.8|20/3.5
compand=points=-80/-80|-9/-9|0/-5.3|20/2.9
compand=points=-80/-80|-12/-12|0/-6.8|20/1.9
compand=points=-80/-80|-18/-18|0/-9.8|20/0.7
compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2
compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6
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
compand=attacks=0:points=-80/-80|-6/-6|20/-6
compand=attacks=0:points=-80/-80|-12/-12|20/-12
compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20
compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8
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:
Set millimeters distance. This is compensation distance for fine tuning. Default is 0.
Set cm distance. This is compensation distance for tightening distance setup. Default is 0.
Set meters distance. This is compensation distance for hard distance setup. Default is 0.
Set dry amount. Amount of unprocessed (dry) signal. Default is 0.
Set wet amount. Amount of processed (wet) signal. Default is 1.
Set temperature degree in Celsius. This is the temperature of the environment. Default is 20.
Apply headphone crossfeed filter.
Crossfeed is the process of blending the left and right channels of stereo audio recording. It is mainly used to reduce extreme stereo separation of low frequencies.
The intent is to produce more speaker like sound to the listener.
The filter accepts the following options:
Set strength of crossfeed. Default is 0.2. Allowed range is from 0 to 1. This sets gain of low shelf filter for side part of stereo image. Default is -6dB. Max allowed is -30db when strength is set to 1.
Set soundstage wideness. Default is 0.5. Allowed range is from 0 to 1. This sets cut off frequency of low shelf filter. Default is cut off near 1550 Hz. With range set to 1 cut off frequency is set to 2100 Hz.
Set input gain. Default is 0.9.
Set output gain. Default is 1.
Simple algorithm to expand audio dynamic range.
The filter accepts the following options:
Sets the intensity of effect (default: 2.0). Must be in range between 0.0 (unchanged sound) to 10.0 (maximum effect).
Enable clipping. By default is enabled.
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.
Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift the audio.
Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to prevent clipping.
Measure audio dynamic range.
DR values of 14 and higher is found in very dynamic material. DR of 8 to 13 is found in transition material. And anything less that 8 have very poor dynamics and is very compressed.
The filter accepts the following options:
Set window length in seconds used to split audio into segments of equal length. Default is 3 seconds.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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:
Set the filter’s central frequency in Hz.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Specify the band-width of a filter in width_type units.
Set the required gain or attenuation in dB. Beware of clipping when using a positive gain.
Specify which channels to filter, by default all available are filtered.
equalizer=f=1000:t=h:width=200:g=-10
equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5
This filter supports the following commands:
Change equalizer frequency. Syntax for the command is : "frequency"
Change equalizer width_type. Syntax for the command is : "width_type"
Change equalizer width. Syntax for the command is : "width"
Change equalizer gain. Syntax for the command is : "gain"
Linearly increases the difference between left and right channels which adds some sort of "live" effect to playback.
The filter accepts the following options:
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.
Enable clipping. By default is enabled.
Apply FIR Equalization using arbitrary frequency response.
The filter accepts the following option:
Set gain curve equation (in dB). The expression can contain variables:
the evaluated frequency
sample rate
channel number, set to 0 when multichannels evaluation is disabled
channel id, see libavutil/channel_layout.h, set to the first channel id when multichannels evaluation is disabled
number of channels
channel_layout, see libavutil/channel_layout.h
and functions:
interpolate gain on frequency f based on gain_entry
same as gain_interpolate, but smoother
This option is also available as command. Default is gain_interpolate(f).
Set gain entry for gain_interpolate function. The expression can contain functions:
store gain entry at frequency f with value g
This option is also available as command.
Set filter delay in seconds. Higher value means more accurate.
Default is 0.01.
Set filter accuracy in Hz. Lower value means more accurate.
Default is 5.
Set window function. Acceptable values are:
rectangular window, useful when gain curve is already smooth
hann window (default)
hamming window
blackman window
3-terms continuous 1st derivative nuttall window
minimum 3-terms discontinuous nuttall window
4-terms continuous 1st derivative nuttall window
minimum 4-terms discontinuous nuttall (blackman-nuttall) window
blackman-harris window
tukey window
If enabled, use fixed number of audio samples. This improves speed when filtering with large delay. Default is disabled.
Enable multichannels evaluation on gain. Default is disabled.
Enable zero phase mode by subtracting timestamp to compensate delay. Default is disabled.
Set scale used by gain. Acceptable values are:
linear frequency, linear gain
linear frequency, logarithmic (in dB) gain (default)
logarithmic (in octave scale where 20 Hz is 0) frequency, linear gain
logarithmic frequency, logarithmic gain
Set file for dumping, suitable for gnuplot.
Set scale for dumpfile. Acceptable values are same with scale option. Default is linlog.
Enable 2-channel convolution using complex FFT. This improves speed significantly. Default is disabled.
Enable minimum phase impulse response. Default is disabled.
firequalizer=gain='if(lt(f,1000), 0, -INF)'
firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
firequalizer=delay=0.1:fixed=on:zero_phase=on
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
Apply a flanging effect to the audio.
The filter accepts the following options:
Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
Set added sweep delay in milliseconds. Range from 0 to 10. Default value is 2.
Set percentage regeneration (delayed signal feedback). Range from -95 to 95. Default value is 0.
Set percentage of delayed signal mixed with original. Range from 0 to 100. Default value is 71.
Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
Set swept wave shape, can be triangular or sinusoidal. Default value is sinusoidal.
Set swept wave percentage-shift for multi channel. Range from 0 to 100. Default value is 25.
Set delay-line interpolation, linear or quadratic. Default is linear.
Apply Haas effect to audio.
Note that this makes most sense to apply on mono signals. With this filter applied to mono signals it give some directionality and stretches its stereo image.
The filter accepts the following options:
Set input level. By default is 1, or 0dB
Set output level. By default is 1, or 0dB.
Set gain applied to side part of signal. By default is 1.
Set kind of middle source. Can be one of the following:
Pick left channel.
Pick right channel.
Pick middle part signal of stereo image.
Pick side part signal of stereo image.
Change middle phase. By default is disabled.
Set left channel delay. By default is 2.05 milliseconds.
Set left channel balance. By default is -1.
Set left channel gain. By default is 1.
Change left phase. By default is disabled.
Set right channel delay. By defaults is 2.12 milliseconds.
Set right channel balance. By default is 1.
Set right channel gain. By default is 1.
Change right phase. By default is enabled.
Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with embedded HDCD codes is expanded into a 20-bit PCM stream.
The filter supports the Peak Extend and Low-level Gain Adjustment features of HDCD, and detects the Transient Filter flag.
ffmpeg -i HDCD16.flac -af hdcd OUT24.flac
When using the filter with wav, note the default encoding for wav is 16-bit,
so the resulting 20-bit stream will be truncated back to 16-bit. Use something
like -acodec pcm_s24le after the filter to get 24-bit PCM output.
ffmpeg -i HDCD16.wav -af hdcd OUT16.wav ffmpeg -i HDCD16.wav -af hdcd -c:a pcm_s24le OUT24.wav
The filter accepts the following options:
Disable any automatic format conversion or resampling in the filter graph.
Process the stereo channels together. If target_gain does not match between channels, consider it invalid and use the last valid target_gain.
Set the code detect timer period in ms.
Always extend peaks above -3dBFS even if PE isn’t signaled.
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:
Disabled
Gain adjustment level at each sample
Samples where peak extend occurs
Samples where the code detect timer is active
Samples where the target gain does not match between channels
Apply head-related transfer functions (HRTFs) to create virtual loudspeakers around the user for binaural listening via headphones. The HRIRs are provided via additional streams, for each channel one stereo input stream is needed.
The filter accepts the following options:
Set mapping of input streams for convolution. The argument is a ’|’-separated list of channel names in order as they are given as additional stream inputs for filter. This also specify number of input streams. Number of input streams must be not less than number of channels in first stream plus one.
Set gain applied to audio. Value is in dB. Default is 0.
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.
Set custom gain for LFE channels. Value is in dB. Default is 0.
Set size of frame in number of samples which will be processed at once. Default value is 1024. Allowed range is from 1024 to 96000.
Set format of hrir stream. Default value is stereo. Alternative value is multich. If value is set to stereo, number of additional streams should be greater or equal to number of input channels in first input stream. Also each additional stream should have stereo number of channels. If value is set to multich, number of additional streams should be exactly one. Also number of input channels of additional stream should be equal or greater than twice number of channels of first input stream.
ffmpeg -i input.wav -lavfi-complex "amovie=azi_270_ele_0_DFC.wav[sr],amovie=azi_90_ele_0_DFC.wav[sl],amovie=azi_225_ele_0_DFC.wav[br],amovie=azi_135_ele_0_DFC.wav[bl],amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe],amovie=azi_35_ele_0_DFC.wav[fl],amovie=azi_325_ele_0_DFC.wav[fr],[a:0][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR" output.wav
ffmpeg -i input.wav -lavfi-complex "amovie=minp.wav[hrirs],[a:0][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich" output.wav
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:
Set frequency in Hz. Default is 3000.
Set number of poles. Default is 2.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
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.
Specify which channels to filter, by default all available are filtered.
This filter supports the following commands:
Change highpass frequency. Syntax for the command is : "frequency"
Change highpass width_type. Syntax for the command is : "width_type"
Change highpass width. Syntax for the command is : "width"
Join multiple input streams into one multi-channel stream.
It accepts the following parameters:
The number of input streams. It defaults to 2.
The desired output channel layout. It defaults to stereo.
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
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.
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/’.
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.
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.
Specify the sample rate, default to 44100. Only used if plugin have zero inputs.
Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.
Set the minimum duration of the sourced audio. See (ffmpeg-utils)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.
ladspa=file=amp
vcf_notch
plugin from VCF library:
ladspa=f=vcf:p=vcf_notch:c=help
Computer Music Toolkit (CMT)
plugin library:
ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12
ladspa=file=tap_reverb:tap_reverb
ladspa=file=cmt:noise_source_white:c=c0=.2
C* Click - Metronome from the
C* Audio Plugin Suite (CAPS) library:
ladspa=file=caps:Click:c=c1=20'
C* Eq10X2 - Stereo 10-band equaliser effect:
ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2
SWH Plugins collection:
ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2
SWH Plugins collection:
ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
Narrower from the C* Audio Plugin Suite
(CAPS) library:
ladspa=caps:Narrower
C* Audio Plugin Suite (CAPS) library:
ladspa=caps:White:.2
C* Audio Plugin Suite (CAPS) library:
ladspa=caps:Fractal:c=c1=1
VLevel plugin:
ladspa=vlevel-ladspa:vlevel_mono
This filter supports the following commands:
Modify the N-th control value.
If the specified value is not valid, it is ignored and prior one is kept.
EBU R128 loudness normalization. Includes both dynamic and linear normalization modes.
Support for both single pass (livestreams, files) and double pass (files) modes.
This algorithm can target IL, LRA, and maximum true peak. To accurately detect true peaks,
the audio stream will be upsampled to 192 kHz unless the normalization mode is linear.
Use the -ar option or aresample filter to explicitly set an output sample rate.
The filter accepts the following options:
Set integrated loudness target. Range is -70.0 - -5.0. Default value is -24.0.
Set loudness range target. Range is 1.0 - 20.0. Default value is 7.0.
Set maximum true peak. Range is -9.0 - +0.0. Default value is -2.0.
Measured IL of input file. Range is -99.0 - +0.0.
Measured LRA of input file. Range is 0.0 - 99.0.
Measured true peak of input file. Range is -99.0 - +99.0.
Measured threshold of input file. Range is -99.0 - +0.0.
Set offset gain. Gain is applied before the true-peak limiter. Range is -99.0 - +99.0. Default is +0.0.
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.
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.
Set print format for stats. Options are summary, json, or none. Default value is none.
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:
Set frequency in Hz. Default is 500.
Set number of poles. Default is 2.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
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.
Specify which channels to filter, by default all available are filtered.
lowpass=c=LFE
This filter supports the following commands:
Change lowpass frequency. Syntax for the command is : "frequency"
Change lowpass width_type. Syntax for the command is : "width_type"
Change lowpass width. Syntax for the command is : "width"
Load a LV2 (LADSPA Version 2) plugin.
To enable compilation of this filter you need to configure FFmpeg with
--enable-lv2.
Specifies the plugin URI. You may need to escape ’:’.
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).
If ‘controls’ is set to help, all available controls and
their valid ranges are printed.
Specify the sample rate, default to 44100. Only used if plugin have zero inputs.
Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.
Set the minimum duration of the sourced audio. See (ffmpeg-utils)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.
lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2
lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5
lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3
Multiband Compress or expand the audio’s dynamic range.
The input audio is divided into bands using 4th order Linkwitz-Riley IIRs. This is akin to the crossover of a loudspeaker, and results in flat frequency response when absent compander action.
It accepts the following parameters:
This option syntax is: attack,decay,[attack,decay..] soft-knee points crossover_frequency [delay [initial_volume [gain]]] | attack,decay ... For explanation of each item refer to compand filter documentation.
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|..."
output channel layout or number of channels
output channel specification, of the form: "out_name=[gain*]in_name[(+-)[gain*]in_name...]"
output channel to define, either a channel name (FL, FR, etc.) or a channel number (c0, c1, etc.)
multiplicative coefficient for the channel, 1 leaving the volume unchanged
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.
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.
The channel remapping will be effective if, and only if:
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 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.
Convert the audio sample format, sample rate and channel layout. It is not meant to be used directly.
Apply time-stretching and pitch-shifting with librubberband.
The filter accepts the following options:
Set tempo scale factor.
Set pitch scale factor.
Set transients detector. Possible values are:
Set detector. Possible values are:
Set phase. Possible values are:
Set processing window size. Possible values are:
Set smoothing. Possible values are:
Enable formant preservation when shift pitching. Possible values are:
Set pitch quality. Possible values are:
Set channels. Possible values are:
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:
Set input gain. Default is 1. Range is between 0.015625 and 64.
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.
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.
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.
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.
Set the amount by how much signal will be amplified after processing. Default is 1. Range is from 1 to 64.
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.
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.
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.
Set sidechain gain. Default is 1. Range is between 0.015625 and 64.
How much to use compressed signal in output. Default is 1. Range is between 0 and 1.
ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"
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:
Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1.
If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.
Set a ratio about which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
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.
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.
Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.
Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak or rms.
Choose if the average level between all channels or the louder channel affects the reduction. Default is average. Can be average or maximum.
Set sidechain gain. Default is 1. Range is from 0.015625 to 64.
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:
Set silence duration until notification (default is 2 seconds).
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.
silencedetect=n=-50dB:d=5
ffmpeg to detect silence with 0.0001 noise
tolerance in ‘silence.mp3’:
ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
Remove silence from the beginning, middle or end of the audio.
The filter accepts the following options:
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.
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.
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.
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.
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.
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.
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.
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.
Set ratio used to calculate size of window for detecting silence.
Default value is 0.02. Allowed range is from 0 to 10.
silenceremove=1:5:0.02
silenceremove=0:0:0:-1:1:-90dB
SOFAlizer uses head-related transfer functions (HRTFs) to create virtual loudspeakers around the user for binaural listening via headphones (audio formats up to 9 channels supported). The HRTFs are stored in SOFA files (see http://www.sofacoustics.org/ for a database). SOFAlizer is developed at the Acoustics Research Institute (ARI) of the Austrian Academy of Sciences.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libmysofa.
The filter accepts the following options:
Set the SOFA file used for rendering.
Set gain applied to audio. Value is in dB. Default is 0.
Set rotation of virtual loudspeakers in deg. Default is 0.
Set elevation of virtual speakers in deg. Default is 0.
Set distance in meters between loudspeakers and the listener with near-field HRTFs. Default is 1.
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.
Set custom positions of virtual loudspeakers. Syntax for this option is: <CH> <AZIM> <ELEV>[|<CH> <AZIM> <ELEV>|...]. Each virtual loudspeaker is described with short channel name following with azimuth and elevation in degrees. Each virtual loudspeaker description is separated by ’|’. For example to override front left and front right channel positions use: ’speakers=FL 45 15|FR 345 15’. Descriptions with unrecognised channel names are ignored.
Set custom gain for LFE channels. Value is in dB. Default is 0.
sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1
sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5
"sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"
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:
Set input level before filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.
Set output level after filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.
Set input balance between both channels. Default is 0. Allowed range is from -1 to 1.
Set output balance between both channels. Default is 0. Allowed range is from -1 to 1.
Enable softclipping. Results in analog distortion instead of harsh digital 0dB clipping. Disabled by default.
Mute the left channel. Disabled by default.
Mute the right channel. Disabled by default.
Change the phase of the left channel. Disabled by default.
Change the phase of the right channel. Disabled by default.
Set stereo mode. Available values are:
Left/Right to Left/Right, this is default.
Left/Right to Mid/Side.
Mid/Side to Left/Right.
Left/Right to Left/Left.
Left/Right to Right/Right.
Left/Right to Left + Right.
Left/Right to Right/Left.
Mid/Side to Left/Left.
Mid/Side to Right/Right.
Set level of side signal. Default is 1. Allowed range is from 0.015625 to 64.
Set balance of side signal. Default is 0. Allowed range is from -1 to 1.
Set level of the middle signal. Default is 1. Allowed range is from 0.015625 to 64.
Set middle signal pan. Default is 0. Allowed range is from -1 to 1.
Set stereo base between mono and inversed channels. Default is 0. Allowed range is from -1 to 1.
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.
Set S/C level. Default is 1. Allowed range is from 1 to 100.
Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.
Set balance mode for balance_in/balance_out option.
Can be one of the following:
Classic balance mode. Attenuate one channel at time. Gain is raised up to 1.
Similar as classic mode above but gain is raised up to 2.
Equal power distribution, from -6dB to +6dB range.
stereotools=mlev=0.015625
"stereotools=mode=ms>lr"
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:
Time in milliseconds of the delay of left signal into right and vice versa. Default is 20 milliseconds.
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.
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.
Set level of input signal of original channel. Default is 0.8.
Apply 18 band equalizer.
The filter accepts the following options:
Set 65Hz band gain.
Set 92Hz band gain.
Set 131Hz band gain.
Set 185Hz band gain.
Set 262Hz band gain.
Set 370Hz band gain.
Set 523Hz band gain.
Set 740Hz band gain.
Set 1047Hz band gain.
Set 1480Hz band gain.
Set 2093Hz band gain.
Set 2960Hz band gain.
Set 4186Hz band gain.
Set 5920Hz band gain.
Set 8372Hz band gain.
Set 11840Hz band gain.
Set 16744Hz band gain.
Set 20000Hz band gain.
Apply audio surround upmix filter.
This filter allows to produce multichannel output from audio stream.
The filter accepts the following options:
Set output channel layout. By default, this is 5.1.
See (ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.
Set input channel layout. By default, this is stereo.
See (ffmpeg-utils)the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.
Set input volume level. By default, this is 1.
Set output volume level. By default, this is 1.
Enable LFE channel output if output channel layout has it. By default, this is enabled.
Set LFE low cut off frequency. By default, this is 128 Hz.
Set LFE high cut off frequency. By default, this is 256 Hz.
Set front center input volume. By default, this is 1.
Set front center output volume. By default, this is 1.
Set LFE input volume. By default, this is 1.
Set LFE output volume. By default, this is 1.
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:
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.
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.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Determine how steep is the filter’s shelf transition.
Specify which channels to filter, by default all available are filtered.
This filter supports the following commands:
Change treble frequency. Syntax for the command is : "frequency"
Change treble width_type. Syntax for the command is : "width_type"
Change treble width. Syntax for the command is : "width"
Change treble gain. Syntax for the command is : "gain"
Sinusoidal amplitude modulation.
The filter accepts the following options:
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.
Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.
Sinusoidal phase modulation.
The filter accepts the following options:
Modulation frequency in Hertz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.
Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.
Adjust the input audio volume.
It accepts the following parameters:
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".
This parameter represents the mathematical precision.
It determines which input sample formats will be allowed, which affects the precision of the volume scaling.
8-bit fixed-point; this limits input sample format to U8, S16, and S32.
32-bit floating-point; this limits input sample format to FLT. (default)
64-bit floating-point; this limits input sample format to DBL.
Choose the behaviour on encountering ReplayGain side data in input frames.
Remove ReplayGain side data, ignoring its contents (the default).
Ignore ReplayGain side data, but leave it in the frame.
Prefer the track gain, if present.
Prefer the album gain, if present.
Pre-amplification gain in dB to apply to the selected replaygain gain.
Default value for replaygain_preamp is 0.0.
Set when the volume expression is evaluated.
It accepts the following values:
only evaluate expression once during the filter initialization, or when the ‘volume’ command is sent
evaluate expression for each incoming frame
Default value is ‘once’.
The volume expression can contain the following parameters.
frame number (starting at zero)
number of channels
number of samples consumed by the filter
number of samples in the current frame
original frame position in the file
frame PTS
sample rate
PTS at start of stream
time at start of stream
frame time
timestamp timebase
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.
This filter supports the following commands:
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.
Prevent clipping by limiting the gain applied.
Default value for replaygain_noclip is 1.
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
volume=volume=6dB:precision=fixed
volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame
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.
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:
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.
Below is a description of the currently available audio sources.
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:
The timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form.
The sample rate of the incoming audio buffers.
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’
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’
The number of channels of the incoming audio buffers. If both channels and channel_layout are specified, then they must be consistent.
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
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:
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.
Set the channel layout. The number of channels in the specified layout must be equal to the number of specified expressions.
Set the minimum duration of the sourced audio. See (ffmpeg-utils)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.
Set the number of samples per channel per each output frame, default to 1024.
Specify the sample rate, default to 44100.
Each expression in exprs can contain the following constants:
number of the evaluated sample, starting from 0
time of the evaluated sample expressed in seconds, starting from 0
sample rate
aevalsrc=0
aevalsrc="sin(440*2*PI*t):s=8000"
aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"
aevalsrc="-2+random(0)"
aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"
aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"
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:
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.
Specifies the sample rate, and defaults to 44100.
Set the number of samples per requested frames.
anullsrc=r=48000:cl=4
anullsrc=r=48000:cl=mono
All the parameters need to be explicitly defined.
Synthesize a voice utterance using the libflite library.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libflite.
Note that versions of the flite library prior to 2.0 are not thread-safe.
The filter accepts the following options:
If set to 1, list the names of the available voices and exit immediately. Default value is 0.
Set the maximum number of samples per frame. Default value is 512.
Set the filename containing the text to speak.
Set the text to speak.
Set the voice to use for the speech synthesis. Default value is
kal. See also the list_voices option.
flite=textfile=speech.txt
slt voice:
flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
flite and
the lavfi device:
ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'
For more information about libflite, check: http://www.festvox.org/flite/
Generate a noise audio signal.
The filter accepts the following options:
Specify the sample rate. Default value is 48000 Hz.
Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value is 1.0.
Specify the duration of the generated audio stream. Not specifying this option results in noise with an infinite length.
Specify the color of noise. Available noise colors are white, pink, brown, blue and violet. Default color is white.
Specify a value used to seed the PRNG.
Set the number of samples per each output frame, default is 1024.
anoisesrc=d=60:c=pink:r=44100:a=0.5
Generate odd-tap Hilbert transform FIR coefficients.
The resulting stream can be used with afir filter for phase-shifting the signal by 90 degrees.
This is used in many matrix coding schemes and for analytic signal generation. The process is often written as a multiplication by i (or j), the imaginary unit.
The filter accepts the following options:
Set sample rate, default is 44100.
Set length of FIR filter, default is 22051.
Set number of samples per each frame.
Set window function to be used when generating FIR coefficients.
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:
Set the carrier frequency. Default is 440 Hz.
Enable a periodic beep every second with frequency beep_factor times the carrier frequency. Default is 0, meaning the beep is disabled.
Specify the sample rate, default is 44100.
Specify the duration of the generated audio stream.
Set the number of samples per output frame.
The expression can contain the following constants:
The (sequential) number of the output audio frame, starting from 0.
The PTS (Presentation TimeStamp) of the output audio frame, expressed in TB units.
The PTS of the output audio frame, expressed in seconds.
The timebase of the output audio frames.
Default is 1024.
sine
sine=220:4:d=5 sine=f=220:b=4:d=5 sine=frequency=220:beep_factor=4:duration=5
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)))'
Below is a description of the currently available audio sinks.
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.
Null audio sink; do absolutely nothing with the input audio. It is mainly useful as a template and for use in analysis / debugging tools.
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.
Extract the alpha component from the input as a grayscale video. This is especially useful with the alphamerge filter.
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.
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:
Set the shaping engine
Available values are:
The default libass shaping engine, which is the best available.
Fast, font-agnostic shaper that can do only substitutions
Slower shaper using OpenType for substitutions and positioning
The default is auto.
Apply an Adaptive Temporal Averaging Denoiser to the video input.
The filter accepts the following options:
Set threshold A for 1st plane. Default is 0.02. Valid range is 0 to 0.3.
Set threshold B for 1st plane. Default is 0.04. Valid range is 0 to 5.
Set threshold A for 2nd plane. Default is 0.02. Valid range is 0 to 0.3.
Set threshold B for 2nd plane. Default is 0.04. Valid range is 0 to 5.
Set threshold A for 3rd plane. Default is 0.02. Valid range is 0 to 0.3.
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.
Set number of frames filter will use for averaging. Default is 33. Must be odd number in range [5, 129].
Set what planes of frame filter will use for averaging. Default is all.
Apply average blur filter.
The filter accepts the following options:
Set horizontal kernel size.
Set which planes to filter. By default all planes are filtered.
Set vertical kernel size, if zero it will be same as sizeX.
Default is 0.
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:
Set the minimal luminance value. Default is 16.
Show and measure bit plane noise.
The filter accepts the following options:
Set which plane to analyze. Default is 1.
Filter out noisy pixels from bitplane set above.
Default is disabled.
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:
Set the minimum detected black duration expressed in seconds. It must be a non-negative floating point number.
Default value is 2.0.
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.
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
Detect frames that are (almost) completely black. Can be useful to detect chapter transitions or commercials. Output lines consist of the frame number of the detected frame, the percentage of blackness, the position in the file if known or -1 and the timestamp in seconds.
In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.
This filter exports frame metadata lavfi.blackframe.pblack.
The value represents the percentage of pixels in the picture that
are below the threshold value.
It accepts the following parameters:
The percentage of the pixels that have to be below the threshold; it defaults to
98.
The threshold below which a pixel value is considered black; it defaults to
32.
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.
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:
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.
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:
The sequential number of the filtered frame, starting from 0.
the coordinates of the current sample
the width and height of currently filtered plane
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.
Time of the current frame, expressed in seconds.
Value of pixel component at current location for first video frame (top layer).
Value of pixel component at current location for second video frame (bottom layer).
The blend filter also supports the framesync options.
blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'
blend=all_expr='A*(X/W)+B*(1-X/W)'
blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'
blend=all_expr='if(gte(N*SW+X,W),A,B)'
blend=all_expr='if(gte(Y-N*SH,0),A,B)'
blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'
blend=all_expr='if(gt(X,Y*(W/H)),A,B)'
tblend=all_mode=grainextract
Apply a boxblur algorithm to the input video.
It accepts the following parameters:
A description of the accepted options follows.
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:
The input width and height in pixels.
The input chroma image width and height in pixels.
The horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p", hsub is 2 and vsub is 1.
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.
boxblur=luma_radius=2:luma_power=1 boxblur=2:1
boxblur=2:1:cr=0:ar=0
boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1
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:
The interlacing mode to adopt. It accepts one of the following values:
Output one frame for each frame.
Output one frame for each field.
The default value is send_field.
The picture field parity assumed for the input interlaced video. It accepts one of the following values:
Assume the top field is first.
Assume the bottom field is first.
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.
Specify which frames to deinterlace. Accept one of the following values:
Deinterlace all frames.
Only deinterlace frames marked as interlaced.
The default value is all.
YUV colorspace color/chroma keying.
The filter accepts the following options:
The color which will be replaced with transparency.
Similarity percentage with the key color.
0.01 matches only the exact key color, while 1.0 matches everything.
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.
Signals that the color passed is already in YUV instead of RGB.
Literal colors like "green" or "red" don’t make sense with this enabled anymore. This can be used to pass exact YUV values as hexadecimal numbers.
ffmpeg -i input.png -vf chromakey=green out.png
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
Display CIE color diagram with pixels overlaid onto it.
The filter accepts the following options:
Set color system.
Set CIE system.
Set what gamuts to draw.
See system option for available values.
Set ciescope size, by default set to 512.
Set intensity used to map input pixel values to CIE diagram.
Set contrast used to draw tongue colors that are out of active color system gamut.
Correct gamma displayed on scope, by default enabled.
Show white point on CIE diagram, by default disabled.
Set input gamma. Used only with XYZ input color space.
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:
Set motion vectors to visualize.
Available flags for mv are:
forward predicted MVs of P-frames
forward predicted MVs of B-frames
backward predicted MVs of B-frames
Display quantization parameters using the chroma planes.
Set motion vectors type to visualize. Includes MVs from all frames unless specified by frame_type option.
Available flags for mv_type are:
forward predicted MVs
backward predicted MVs
Set frame type to visualize motion vectors of.
Available flags for frame_type are:
intra-coded frames (I-frames)
predicted frames (P-frames)
bi-directionally predicted frames (B-frames)
ffplay:
ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv_type=fp
ffplay:
ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv=pf+bf+bb
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:
Adjust red, green and blue shadows (darkest pixels).
Adjust red, green and blue midtones (medium pixels).
Adjust red, green and blue highlights (brightest pixels).
Allowed ranges for options are [-1.0, 1.0]. Defaults are 0.
colorbalance=rs=.3
RGB colorspace color keying.
The filter accepts the following options:
The color which will be replaced with transparency.
Similarity percentage with the key color.
0.01 matches only the exact key color, while 1.0 matches everything.
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.
ffmpeg -i input.png -vf colorkey=green out.png
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
Adjust video input frames using levels.
The filter accepts the following options:
Adjust red, green, blue and alpha input black point.
Allowed ranges for options are [-1.0, 1.0]. Defaults are 0.
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.
Adjust red, green, blue and alpha output black point.
Allowed ranges for options are [0, 1.0]. Defaults are 0.
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.
colorlevels=rimin=0.058:gimin=0.058:bimin=0.058
colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96
colorlevels=rimax=0.902:gimax=0.902:bimax=0.902
colorlevels=romin=0.5:gomin=0.5:bomin=0.5
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:
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.
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.
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.
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].
colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3
colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131
Convert color matrix.
The filter accepts the following options:
Specify the source and destination color matrix. Both values must be specified.
The accepted values are:
BT.709
FCC
BT.601
BT.470
BT.470BG
SMPTE-170M
SMPTE-240M
BT.2020
For example to convert from BT.601 to SMPTE-240M, use the command:
colormatrix=bt601:smpte240m
Convert colorspace, transfer characteristics or color primaries. Input video needs to have an even size.
The filter accepts the following options:
Specify all color properties at once.
The accepted values are:
BT.470M
BT.470BG
BT.601-6 525
BT.601-6 625
BT.709
SMPTE-170M
SMPTE-240M
BT.2020
Specify output colorspace.
The accepted values are:
BT.709
FCC
BT.470BG or BT.601-6 625
SMPTE-170M or BT.601-6 525
SMPTE-240M
YCgCo
BT.2020 with non-constant luminance
Specify output transfer characteristics.
The accepted values are:
BT.709
BT.470M
BT.470BG
Constant gamma of 2.2
Constant gamma of 2.8
SMPTE-170M, BT.601-6 625 or BT.601-6 525
SMPTE-240M
SRGB
iec61966-2-1
iec61966-2-4
xvycc
BT.2020 for 10-bits content
BT.2020 for 12-bits content
Specify output color primaries.
The accepted values are:
BT.709
BT.470M
BT.470BG or BT.601-6 625
SMPTE-170M or BT.601-6 525
SMPTE-240M
film
SMPTE-431
SMPTE-432
BT.2020
JEDEC P22 phosphors
Specify output color range.
The accepted values are:
TV (restricted) range
MPEG (restricted) range
PC (full) range
JPEG (full) range
Specify output color format.
The accepted values are:
YUV 4:2:0 planar 8-bits
YUV 4:2:0 planar 10-bits
YUV 4:2:0 planar 12-bits
YUV 4:2:2 planar 8-bits
YUV 4:2:2 planar 10-bits
YUV 4:2:2 planar 12-bits
YUV 4:4:4 planar 8-bits
YUV 4:4:4 planar 10-bits
YUV 4:4:4 planar 12-bits
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.
Specify dithering mode.
The accepted values are:
No dithering
Floyd-Steinberg dithering
Whitepoint adaptation mode.
The accepted values are:
Bradford whitepoint adaptation
von Kries whitepoint adaptation
identity whitepoint adaptation (i.e. no whitepoint adaptation)
Override all input properties at once. Same accepted values as all.
Override input colorspace. Same accepted values as space.
Override input color primaries. Same accepted values as primaries.
Override input transfer characteristics. Same accepted values as trc.
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
Apply convolution 3x3, 5x5 or 7x7 filter.
The filter accepts the following options:
Set matrix for each plane. Matrix is sequence of 9, 25 or 49 signed integers.
Set multiplier for calculated value for each plane.
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.
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"
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"
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"
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"
convolution="1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0"
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"
Apply 2D convolution of video stream in frequency domain using second stream as impulse.
The filter accepts the following options:
Set which planes to process.
Set which impulse video frames will be processed, can be first or all. Default is all.
The convolve filter also supports the framesync options.
Copy the input video source unchanged to the output. This is mainly useful for testing purposes.
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 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
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>][@...]][#...]
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.
coreimage=list_filters=true
coreimage=filter=CIBoxBlur@default
coreimage=filter=CIBoxBlur@default#CIVignetteEffect@inputCenter=100\ 100@inputRadius=50
ffmpeg -f lavfi -i nullsrc=s=100x100,coreimage=filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png
Crop the input video to given dimensions.
It accepts the following parameters:
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.
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.
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.
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.
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.
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:
The computed values for x and y. They are evaluated for each new frame.
The input width and height.
These are the same as in_w and in_h.
The output (cropped) width and height.
These are the same as out_w and out_h.
same as iw / ih
input sample aspect ratio
input display aspect ratio, it is the same as (iw / ih) * sar
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
The number of the input frame, starting from 0.
the position in the file of the input frame, NAN if unknown
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.
crop=100:100:12:34
Using named options, the example above becomes:
crop=w=100:h=100:x=12:y=34
crop=100:100
crop=2/3*in_w:2/3*in_h
crop=out_w=in_h crop=in_h
crop=in_w-100:in_h-100:100:100
crop=in_w-2*10:in_h-2*20
crop=in_w/2:in_h/2:in_w/2:in_h/2
crop=in_w:1/PHI*in_w
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)
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)"
crop=in_w/2:in_h/2:y:10+10*sin(n/10)
This filter supports the following commands:
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.
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:
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.
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.
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.
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:
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:
Default is none.
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.
Set the key points for the red component.
Set the key points for the green component.
Set the key points for the blue component.
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.
Specify a Photoshop curves file (.acv) to import the settings from.
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 ....
curves=blue='0/0 0.5/0.58 1/1'
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:
(0;0.11) (0.42;0.51) (1;0.95)
(0;0) (0.50;0.48) (1;1)
(0;0.22) (0.49;0.44) (1;0.80)
curves=preset=vintage
curves=vintage
curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'
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
Video data analysis filter.
This filter shows hexadecimal pixel values of part of video.
The filter accepts the following options:
Set output video size.
Set x offset from where to pick pixels.
Set y offset from where to pick pixels.
Set scope mode, can be one of the following:
Draw hexadecimal pixel values with white color on black background.
Draw hexadecimal pixel values with input video pixel color on black background.
Draw hexadecimal pixel values on color background picked from input video, the text color is picked in such way so its always visible.
Draw rows and columns numbers on left and top of video.
Set background opacity.
Denoise frames using 2D DCT (frequency domain filtering).
This filter is not designed for real time.
The filter accepts the following options:
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.
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.
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.
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.
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
Remove banding artifacts from input video. It works by replacing banded pixels with average value of referenced pixels.
The filter accepts the following options:
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.
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.
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.
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.
If enabled, current pixel is changed if and only if all pixel components are banded, e.g. banding detection threshold is triggered for all color components. The default is disabled.
Drop duplicated frames at regular intervals.
The filter accepts the following options:
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.
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
Set scene change threshold. Default is 15.
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.
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.
Set whether or not chroma is considered in the metric calculations. Default is
1.
Apply 2D deconvolution of video stream in frequency domain using second stream as impulse.
The filter accepts the following options:
Set which planes to process.
Set which impulse video frames will be processed, can be first or all. Default is all.
Set noise when doing divisions. Default is 0.0000001. Useful when width and height are not same and not power of 2 or if stream prior to convolving had noise.
The deconvolve filter also supports the framesync options.
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:
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
Remove temporal frame luminance variations.
It accepts the following options:
Set moving-average filter size in frames. Default is 5. Allowed range is 2 - 129.
Set averaging mode to smooth temporal luminance variations.
Available values are:
Arithmetic mean
Geometric mean
Harmonic mean
Quadratic mean
Cubic mean
Power mean
Median
Do not actually modify frame. Useful when one only wants metadata.
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:
Specify the length of the window over which the judder repeats.
Accepts any integer greater than 1. Useful values are:
If the original was telecined from 24 to 30 fps (Film to NTSC).
If the original was telecined from 25 to 30 fps (PAL to NTSC).
If a mixture of the two.
The default is ‘4’.
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:
Specify the top left corner coordinates of the logo. They must be specified.
Specify the width and height of the logo to clear. They must be specified.
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.
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.
delogo=x=0:y=0:w=100:h=77:band=10
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:
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.
Specify the maximum extent of movement in x and y directions in the range 0-64 pixels. Default 16.
Specify how to generate pixels to fill blanks at the edge of the frame. Available values are:
Fill zeroes at blank locations
Original image at blank locations
Extruded edge value at blank locations
Mirrored edge at blank locations
Default value is ‘mirror’.
Specify the blocksize to use for motion search. Range 4-128 pixels, default 8.
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.
Specify the search strategy. Available values are:
Set exhaustive search
Set less exhaustive search.
Default value is ‘exhaustive’.
If set then a detailed log of the motion search is written to the specified file.
Remove unwanted contamination of foreground colors, caused by reflected color of greenscreen or bluescreen.
This filter accepts the following options:
Set what type of despill to use.
Set how spillmap will be generated.
Set how much to get rid of still remaining spill.
Controls amount of red in spill area.
Controls amount of green in spill area. Should be -1 for greenscreen.
Controls amount of blue in spill area. Should be -1 for bluescreen.
Controls brightness of spill area, preserving colors.
Modify alpha from generated spillmap.
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:
top field first
bottom field first
The default value is top.
A string of numbers representing the pulldown pattern you wish to apply.
The default value is 23.
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.
Apply dilation effect to the video.
This filter replaces the pixel by the local(3x3) maximum.
It accepts the following options:
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
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 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.
Set displace behavior for pixels that are out of range.
Available values are:
Missing pixels are replaced by black pixels.
Adjacent pixels will spread out to replace missing pixels.
Out of range pixels are wrapped so they point to pixels of other side.
Out of range pixels will be replaced with mirrored pixels.
Default is ‘smear’.
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
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
Draw a colored box on the input image.
It accepts the following parameters:
The expressions which specify the top left corner coordinates of the box. It defaults to 0.
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.
Specify the color of the box to write. For the general syntax of this option,
check the (ffmpeg-utils)"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.
The expression which sets the thickness of the box edge.
A value of fill will create a filled box. Default value is 3.
See below for the list of accepted constants.
Applicable if the input has alpha. With value 1, the pixels of the painted box
will overwrite the video’s color and alpha pixels.
Default is 0, which composites the box onto the input, leaving the video’s alpha intact.
The parameters for x, y, w and h and t are expressions containing the following constants:
The input display aspect ratio, it is the same as (w / h) * sar.
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
The input width and height.
The input sample aspect ratio.
The x and y offset coordinates where the box is drawn.
The width and height of the drawn box.
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.
drawbox
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
drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=fill
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
Draw a grid on the input image.
It accepts the following parameters:
The expressions which specify the coordinates of some point of grid intersection (meant to configure offset). Both default to 0.
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.
Specify the color of the grid. For the general syntax of this option,
check the (ffmpeg-utils)"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.
The expression which sets the thickness of the grid line. Default value is 1.
See below for the list of accepted constants.
Applicable if the input has alpha. With 1 the pixels of the painted grid
will overwrite the video’s color and alpha pixels.
Default is 0, which composites the grid onto the input, leaving the video’s alpha intact.
The parameters for x, y, w and h and t are expressions containing the following constants:
The input display aspect ratio, it is the same as (w / h) * sar.
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
The input grid cell width and height.
The input sample aspect ratio.
The x and y coordinates of some point of grid intersection (meant to configure offset).
The width and height of the drawn cell.
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.
drawgrid=width=100:height=100:thickness=2:color=red@0.5
drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5
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.
It accepts the following parameters:
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.
Set the width of the border to be drawn around the box using boxcolor. The default value of boxborderw is 0.
The color to be used for drawing box around text. For the syntax of this option, check the (ffmpeg-utils)"Color" section in the ffmpeg-utils manual.
The default value of boxcolor is "white".
Set the line spacing in pixels of the border to be drawn around the box using box. The default value of line_spacing is 0.
Set the width of the border to be drawn around the text using bordercolor. The default value of borderw is 0.
Set the color to be used for drawing border around text. For the syntax of this option, check the (ffmpeg-utils)"Color" section in the ffmpeg-utils manual.
The default value of bordercolor is "black".
Select how the text is expanded. Can be either none,
strftime (deprecated) or
normal (default). See the Text expansion section
below for details.
Set a start time for the count. Value is in microseconds. Only applied
in the deprecated strftime expansion mode. To emulate in normal expansion
mode use the pts function, supplying the start time (in seconds)
as the second argument.
If true, check and fix text coords to avoid clipping.
The color to be used for drawing fonts. For the syntax of this option, check the (ffmpeg-utils)"Color" section in the ffmpeg-utils manual.
The default value of fontcolor is "black".
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.
The font family to be used for drawing text. By default Sans.
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 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.
The font size to be used for drawing text. The default value of fontsize is 16.
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).
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 value is "default".
For more information consult the documentation for the FT_LOAD_* libfreetype flags.
The color to be used for drawing a shadow behind the drawn text. For the syntax of this option, check the (ffmpeg-utils)"Color" section in the ffmpeg-utils manual.
The default value of shadowcolor is "black".
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".
The starting frame number for the n/frame_num variable. The default value is "0".
The size in number of spaces to use for rendering the tab. Default value is 4.
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.
Set the timecode frame rate (timecode only). Value will be rounded to nearest integer. Minimum value is "1". Drop-frame timecode is supported for frame rates 30 & 60.
If set to 1, the output of the timecode option will wrap around at 24 hours. Default is 0 (disabled).
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.
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.
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.
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:
input display aspect ratio, it is the same as (w / h) * sar
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
the height of each text line
the input height
the input width
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.
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.
maximum glyph height, that is the maximum height for all the glyphs contained in the rendered text, it is equivalent to ascent - descent.
maximum glyph width, that is the maximum width for all the glyphs contained in the rendered text
the number of input frame, starting from 0
return a random number included between min and max
The input sample aspect ratio.
timestamp expressed in seconds, NAN if the input timestamp is unknown
the height of the rendered text
the width of the rendered text
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.
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, eThe 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, eifEvaluate 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.
gmtimeThe time at which the filter is running, expressed in UTC. It can accept an argument: a strftime() format string.
localtimeThe time at which the filter is running, expressed in the local time zone. It can accept an argument: a strftime() format string.
metadataFrame 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_numThe frame number, starting from 0.
pict_typeA 1 character description of the current picture type.
ptsThe 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.
drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"
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.
drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"
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)"
drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t"
drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t"
drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent"
drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'"
drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'
drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'
#!/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 }"
drawtext=fontfile=FreeSans.ttf:text=DOG:fontsize=24:x=10:y=20+24-max_glyph_a, drawtext=fontfile=FreeSans.ttf:text=cow:fontsize=24:x=80:y=20+24-max_glyph_a
For more information about libfreetype, check: http://www.freetype.org/.
For more information about fontconfig, check: http://freedesktop.org/software/fontconfig/fontconfig-user.html.
For more information about libfribidi, check: http://fribidi.org/.
Detect and draw edges. The filter uses the Canny Edge Detection algorithm.
The filter accepts the following options:
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.
Define the drawing mode.
Draw white/gray wires on black background.
Mix the colors to create a paint/cartoon effect.
Default value is wires.
edgedetect=low=0.1:high=0.4
edgedetect=mode=colormix:high=0
Set brightness, contrast, saturation and approximate gamma adjustment.
The filter accepts the following options:
Set the contrast expression. The value must be a float value in range
-2.0 to 2.0. The default value is "1".
Set the brightness expression. The value must be a float value in
range -1.0 to 1.0. The default value is "0".
Set the saturation expression. The value must be a float in
range 0.0 to 3.0. The default value is "1".
Set the gamma expression. The value must be a float in range
0.1 to 10.0. The default value is "1".
Set the gamma expression for red. The value must be a float in
range 0.1 to 10.0. The default value is "1".
Set the gamma expression for green. The value must be a float in range
0.1 to 10.0. The default value is "1".
Set the gamma expression for blue. The value must be a float in range
0.1 to 10.0. The default value is "1".
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".
Set when the expressions for brightness, contrast, saturation and gamma expressions are evaluated.
It accepts the following values:
only evaluate expressions once during the filter initialization or when a command is processed
evaluate expressions for each incoming frame
Default value is ‘init’.
The expressions accept the following parameters:
frame count of the input frame starting from 0
byte position of the corresponding packet in the input file, NAN if unspecified
frame rate of the input video, NAN if the input frame rate is unknown
timestamp expressed in seconds, NAN if the input timestamp is unknown
The filter supports the following commands:
Set the contrast expression.
Set the brightness expression.
Set the saturation expression.
Set the gamma expression.
Set the gamma_r expression.
Set gamma_g expression.
Set gamma_b expression.
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.
Apply erosion effect to the video.
This filter replaces the pixel by the local(3x3) minimum.
It accepts the following options:
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
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
Extract color channel components from input video stream into separate grayscale video streams.
The filter accepts the following option:
Set plane(s) to extract.
Available values for planes are:
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.
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
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.
Set codebook length. The value must be a positive integer, and represents the number of distinct output colors. Default value is 256.
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.
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.
Set pal8 output pixel format. This option does not work with codebook length greater than 256.
Measure graylevel entropy in histogram of color channels of video frames.
It accepts the following parameters:
Can be either normal or diff. Default is normal.
diff mode measures entropy of histogram delta values, absolute differences between neighbour histogram values.
Apply a fade-in/out effect to the input video.
It accepts the following parameters:
The effect type can be either "in" for a fade-in, or "out" for a fade-out
effect.
Default is in.
Specify the number of the frame to start applying the fade effect at. Default is 0.
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.
If set to 1, fade only alpha channel, if one exists on the input. Default value is 0.
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.
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).
Specify the color of the fade. Default is "black".
fade=in:0:30
The command above is equivalent to:
fade=t=in:s=0:n=30
fade=out:155:45 fade=type=out:start_frame=155:nb_frames=45
fade=in:0:25, fade=out:975:25
fade=in:5:20:color=yellow
fade=in:0:25:alpha=1
fade=t=in:st=5.5:d=0.5
Apply arbitrary expressions to samples in frequency domain
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.
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.
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.
Set the frequency domain weight expression for the luma plane.
Set the frequency domain weight expression for the 1st chroma plane.
Set the frequency domain weight expression for the 2nd chroma plane.
Set when the expressions are evaluated.
It accepts the following values:
Only evaluate expressions once during the filter initialization.
Evaluate expressions for each incoming frame.
Default value is ‘init’.
The filter accepts the following variables:
The coordinates of the current sample.
The width and height of the image.
The number of input frame, starting from 0.
fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'
fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'
fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'
fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))'
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:
Specify whether to extract the top (if the value is 0 or
top) or the bottom field (if the value is 1 or
bottom).
Create new frames by copying the top and bottom fields from surrounding frames supplied as numbers by the hint file.
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.
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 -
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:
Specify the assumed field order of the input stream. Available values are:
Auto detect parity (use FFmpeg’s internal parity value).
Assume bottom field first.
Assume top field first.
Note that it is sometimes recommended not to trust the parity announced by the stream.
Default value is auto.
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:
2-way matching (p/c)
2-way matching, and trying 3rd match if still combed (p/c + n)
2-way matching, and trying 3rd match (same order) if still combed (p/c + u)
2-way matching, trying 3rd match if still combed, and trying 4th/5th matches if still combed (p/c + n + u/b)
3-way matching (p/c/n)
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.
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).
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:
Automatic (same value as ‘order’).
Match from the bottom field.
Match from the top field.
Default value is auto.
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.
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.
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.
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:
No final matching based on combed scores.
Combed scores are only used when a scene change is detected.
Use combed scores all the time.
Default is sc.
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:
No forced calculation.
Force p/c/n calculations.
Force p/c/n/u/b calculations.
Default value is none.
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.
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.
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.
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.
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:
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
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
Transform the field order of the input video.
It accepts the following parameters:
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
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.
Fill borders of the input video, without changing video stream dimensions. Sometimes video can have garbage at the four edges and you may not want to crop video input to keep size multiple of some number.
This filter accepts the following options:
Number of pixels to fill from left border.
Number of pixels to fill from right border.
Number of pixels to fill from top border.
Number of pixels to fill from bottom border.
Set fill mode.
It accepts the following values:
fill pixels using outermost pixels
fill pixels using mirroring
fill pixels with constant value
Default is smear.
Set color for pixels in fixed mode. Default is black.
Find a rectangular object
It accepts the following options:
Filepath of the object image, needs to be in gray8.
Detection threshold, default is 0.5.
Number of mipmaps, default is 3.
Specifies the rectangle in which to search.
ffmpeg:
ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
Cover a rectangular object
It accepts the following options:
Filepath of the optional cover image, needs to be in yuv420.
Set covering mode.
It accepts the following values:
cover it by the supplied image
cover it by interpolating the surrounding pixels
Default value is blur.
ffmpeg:
ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
Flood area with values of same pixel components with another values.
It accepts the following options:
Set pixel x coordinate.
Set pixel y coordinate.
Set source #0 component value.
Set source #1 component value.
Set source #2 component value.
Set source #3 component value.
Set destination #0 component value.
Set destination #1 component value.
Set destination #2 component value.
Set destination #3 component value.
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:
A ’|’-separated list of pixel format names, such as "pix_fmts=yuv420p|monow|rgb24".
format=pix_fmts=yuv420p
Convert the input video to any of the formats in the list
format=pix_fmts=yuv420p|yuv444p|yuv410p
Convert the video to specified constant frame rate by duplicating or dropping frames as necessary.
It accepts the following parameters:
The desired output frame rate. The default is 25.
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.
Timestamp (PTS) rounding method.
Possible values are:
round towards 0
round away from 0
round towards -infinity
round towards +infinity
round to nearest
The default is near.
Action performed when reading the last frame.
Possible values are:
Use same timestamp rounding method as used for other frames.
Pass through last frame if input duration has not been reached yet.
The default is round.
Alternatively, the options can be specified as a flat string: fps[:start_time[:round]].
See also the setpts filter.
fps=fps=25
fps=fps=film:round=near
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:
The desired packing format. Supported values are:
The views are next to each other (default).
The views are on top of each other.
The views are packed by line.
The views are packed by column.
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
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.
Specify the output frames per second. This option can also be specified
as a value alone. The default is 50.
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.
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.
Specify the level at which a scene change is detected as a value between
0 and 100 to indicate a new scene; a low value reflects a low
probability for the current frame to introduce a new scene, while a higher
value means the current frame is more likely to be one.
The default is 8.2.
Specify flags influencing the filter process.
Available value for flags is:
Enable scene change detection using the value of the option scene. This flag is enabled by default.
Select one frame every N-th frame.
This filter accepts the following option:
Select frame after every step frames.
Allowed values are positive integers higher than 0. Default value is 1.
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:
The name of the frei0r effect to load. If the environment variable
FREI0R_PATH is defined, the frei0r effect is searched for in each of the
directories specified by the colon-separated list in FREI0R_PATH.
Otherwise, the standard frei0r paths are searched, in this order:
‘HOME/.frei0r-1/lib/’, ‘/usr/local/lib/frei0r-1/’,
‘/usr/lib/frei0r-1/’.
A ’|’-separated list of parameters to pass to the frei0r effect.
A frei0r effect parameter can be a boolean (its value is either "y" or "n"), a double, a color (specified as R/G/B, where R, G, and B are floating point numbers between 0.0 and 1.0, inclusive) or a color description as specified in the (ffmpeg-utils)"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.
frei0r=filter_name=distort0r:filter_params=0.5|0.01
frei0r=colordistance:0.2/0.3/0.4 frei0r=colordistance:violet frei0r=colordistance:0x112233
frei0r=perspective:0.2/0.2|0.8/0.2
For more information, see http://frei0r.dyne.org
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:
Set quality. This option defines the number of levels for averaging. It accepts
an integer in the range 4-5. Default value is 4.
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).
Set filter strength. It accepts an integer in range -15 to 32. Lower values mean
more details but also more artifacts, while higher values make the image smoother
but also blurrier. Default value is 0 − PSNR optimal.
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).
Apply Gaussian blur filter.
The filter accepts the following options:
Set horizontal sigma, standard deviation of Gaussian blur. Default is 0.5.
Set number of steps for Gaussian approximation. Defauls is 1.
Set which planes to filter. By default all planes are filtered.
Set vertical sigma, if negative it will be same as sigma.
Default is -1.
The filter accepts the following options:
Set the luminance expression.
Set the chrominance blue expression.
Set the chrominance red expression.
Set the alpha expression.
Set the red expression.
Set the green expression.
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:
The sequential number of the filtered frame, starting from 0.
The coordinates of the current sample.
The width and height of the image.
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.
Time of the current frame, expressed in seconds.
Return the value of the pixel at location (x,y) of the current plane.
Return the value of the pixel at location (x,y) of the luminance plane.
Return the value of the pixel at location (x,y) of the blue-difference chroma plane. Return 0 if there is no such plane.
Return the value of the pixel at location (x,y) of the red-difference chroma plane. Return 0 if there is no such plane.
Return the value of the pixel at location (x,y) of the red/green/blue component. Return 0 if there is no such component.
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.
geq=p(W-X\,Y)
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
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
format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'
geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'
geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray
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:
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.
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]
3.5 strength and radius of 8:
gradfun=3.5:8
gradfun=radius=8
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:
Force termination when the shortest input terminates. Default is 0.
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.
Generate an identity Hald CLUT stream altered with various effects:
ffmpeg -f lavfi -i 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.
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 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"
Flip the input video horizontally.
For example, to horizontally flip the input video with ffmpeg:
ffmpeg -i in.avi -vf "hflip" out.avi
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:
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.
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.
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.
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:
Set height of level. Default value is 200.
Allowed range is [50, 2048].
Set height of color scale. Default value is 12.
Allowed range is [0, 40].
Set display mode. It accepts the following values:
Per color component graphs are placed below each other.
Per color component graphs are placed side by side.
Presents information identical to that in the parade, except
that the graphs representing color components are superimposed directly
over one another.
Default is stack.
Set mode. Can be either linear, or logarithmic.
Default is linear.
Set what color components to display.
Default is 7.
Set foreground opacity. Default is 0.7.
Set background opacity. Default is 0.5.
ffplay -i input -vf histogram
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:
A non-negative floating point number which specifies spatial luma strength. It defaults to 4.0.
A non-negative floating point number which specifies spatial chroma strength. It defaults to 3.0*luma_spatial/4.0.
A floating point number which specifies luma temporal strength. It defaults to 6.0*luma_spatial/4.0.
A floating point number which specifies chroma temporal strength. It defaults to luma_tmp*chroma_spatial/luma_spatial.
Download hardware frames to system memory.
The input must be in hardware frames, and the output a non-hardware format. Not all formats will be supported on the output - it may be necessary to insert an additional ‘format’ filter immediately following in the graph to get the output in a supported format.
Map hardware frames to system memory or to another device.
This filter has several different modes of operation; which one is used depends on the input and output formats:
Map the input frames to system memory and pass them to the output. If the original hardware frame is later required (for example, after overlaying something else on part of it), the ‘hwmap’ filter can be used again in the next mode to retrieve it.
If the input is actually a software-mapped hardware frame, then unmap it - that is, return the original hardware frame.
Otherwise, a device must be provided. Create new hardware surfaces on that device for the output, then map them back to the software format at the input and give those frames to the preceding filter. This will then act like the ‘hwupload’ filter, but may be able to avoid an additional copy when the input is already in a compatible format.
A device must be supplied for the output, either directly or with the ‘derive_device’ option. The input and output devices must be of different types and compatible - the exact meaning of this is system-dependent, but typically it means that they must refer to the same underlying hardware context (for example, refer to the same graphics card).
If the input frames were originally created on the output device, then unmap to retrieve the original frames.
Otherwise, map the frames to the output device - create new hardware frames on the output corresponding to the frames on the input.
The following additional parameters are accepted:
Set the frame mapping mode. Some combination of:
The mapped frame should be readable.
The mapped frame should be writeable.
The mapping will always overwrite the entire frame.
This may improve performance in some cases, as the original contents of the frame need not be loaded.
The mapping must not involve any copying.
Indirect mappings to copies of frames are created in some cases where either direct mapping is not possible or it would have unexpected properties. Setting this flag ensures that the mapping is direct and will fail if that is not possible.
Defaults to read+write if not specified.
Rather than using the device supplied at initialisation, instead derive a new device of type type from the device the input frames exist on.
In a hardware to hardware mapping, map in reverse - create frames in the sink and map them back to the source. This may be necessary in some cases where a mapping in one direction is required but only the opposite direction is supported by the devices being used.
This option is dangerous - it may break the preceding filter in undefined ways if there are any additional constraints on that filter’s output. Do not use it without fully understanding the implications of its use.
Upload system memory frames to hardware surfaces.
The device to upload to must be supplied when the filter is initialised. If using ffmpeg, select the appropriate device with the ‘-filter_hw_device’ option.
Upload system memory frames to a CUDA device.
It accepts the following optional parameters:
The number of the CUDA device to use
Apply a high-quality magnification filter designed for pixel art. This filter was originally created by Maxim Stepin.
It accepts the following option:
Set the scaling dimension: 2 for hq2x, 3 for
hq3x and 4 for hq4x.
Default is 3.
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:
Set number of input streams. Default is 2.
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.
Modify the hue and/or the saturation of the input.
It accepts the following parameters:
Specify the hue angle as a number of degrees. It accepts an expression, and defaults to "0".
Specify the saturation in the [-10,10] range. It accepts an expression and defaults to "1".
Specify the hue angle as a number of radians. It accepts an expression, and defaults to "0".
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:
frame count of the input frame starting from 0
presentation timestamp of the input frame expressed in time base units
frame rate of the input video, NAN if the input frame rate is unknown
timestamp expressed in seconds, NAN if the input timestamp is unknown
time base of the input video
hue=h=90:s=1
hue=H=PI/2:s=1
hue="H=2*PI*t: s=sin(2*PI*t)+1"
hue="s=min(t/3\,1)"
The general fade-in expression can be written as:
hue="s=min(0\, max((t-START)/DURATION\, 1))"
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))"
This filter supports the following commands:
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.
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:
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.
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.
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:
Detected type of current frame using single-frame detection. One of: “tff” (top field first), “bff” (bottom field first), “progressive”, or “undetermined”
Cumulative number of frames detected as top field first using single-frame detection.
Cumulative number of frames detected as top field first using multiple-frame detection.
Cumulative number of frames detected as bottom field first using single-frame detection.
Detected type of current frame using multiple-frame detection. One of: “tff” (top field first), “bff” (bottom field first), “progressive”, or “undetermined”
Cumulative number of frames detected as bottom field first using multiple-frame detection.
Cumulative number of frames detected as progressive using single-frame detection.
Cumulative number of frames detected as progressive using multiple-frame detection.
Cumulative number of frames that could not be classified using single-frame detection.
Cumulative number of frames that could not be classified using multiple-frame detection.
Which field in the current frame is repeated from the last. One of “neither”, “top”, or “bottom”.
Cumulative number of frames with no repeated field.
Cumulative number of frames with the top field repeated from the previous frame’s top field.
Cumulative number of frames with the bottom field repeated from the previous frame’s bottom field.
The filter accepts the following options:
Set interlacing threshold.
Set progressive threshold.
Threshold for repeated field detection.
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.
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
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:
Available values for luma_mode, chroma_mode and alpha_mode are:
Do nothing.
Deinterleave fields, placing one above the other.
Interleave fields. Reverse the effect of deinterleaving.
Default value is none.
Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is 0.
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:
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
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:
This determines whether the interlaced frame is taken from the even (tff - default) or odd (bff) lines of the progressive frame.
Vertical lowpass filter to avoid twitter interlacing and reduce moire patterns.
Disable vertical lowpass filter
Enable linear filter (default)
Enable complex filter. This will slightly less reduce twitter and moire but better retain detail and subjective sharpness impression.
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.
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.
Paint pixels exceeding the threshold value to white if set to 1. Default is 0.
Set the fields order. Swap fields if set to 1, leave fields alone if 0. Default is 0.
Enable additional sharpening if set to 1. Default is 0.
Enable twoway sharpening if set to 1. Default is 0.
kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0
kerndeint=sharp=1
kerndeint=map=1
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.
The filter accepts the following options:
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. Default is 0.5.
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. Default is 0.5.
Coefficient of the quadratic correction term. This value has a range [-1,1]. 0 means no correction. Default is 0.
Coefficient of the double quadratic correction term. This value has a range [-1,1]. 0 means no correction. Default is 0.
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.
Obtain the VMAF (Video Multi-Method Assessment Fusion) score between two input videos.
The obtained VMAF score is printed through the logging system.
It requires Netflix’s vmaf library (libvmaf) as a pre-requisite.
After installing the library it can be enabled using:
./configure --enable-libvmaf.
If no model path is specified it uses the default model: vmaf_v0.6.1.pkl.
The filter has following options:
Set the model path which is to be used for SVM.
Default value: "vmaf_v0.6.1.pkl"
Set the file path to be used to store logs.
Set the format of the log file (xml or json).
Enables transform for computing vmaf.
Invokes the phone model which will generate VMAF scores higher than in the regular model, which is more suitable for laptop, TV, etc. viewing conditions.
Enables computing psnr along with vmaf.
Enables computing ssim along with vmaf.
Enables computing ms_ssim along with vmaf.
Set the pool method (mean, min or harmonic mean) to be used for computing vmaf.
This filter also supports the framesync options.
On the below examples the input file ‘main.mpg’ being processed is compared with the reference file ‘ref.mpg’.
ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf -f null -
Example with options:
ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf="psnr=1:enable-transform=1" -f null -
Limits the pixel components values to the specified range [min, max].
The filter accepts the following options:
Lower bound. Defaults to the lowest allowed value for the input.
Upper bound. Defaults to the highest allowed value for the input.
Specify which planes will be processed. Defaults to all available.
Loop video frames.
The filter accepts the following options:
Set the number of loops. Setting this value to -1 will result in infinite loops. Default is 0.
Set maximal size in number of frames. Default is 0.
Set first frame of loop. Default is 0.
Apply a 3D LUT to an input video.
The filter accepts the following options:
Set the 3D LUT file name.
Currently supported formats:
AfterEffects
Iridas
DaVinci
Pandora
Select interpolation mode.
Available values are:
Use values from the nearest defined point.
Interpolate values using the 8 points defining a cube.
Interpolate values using a tetrahedron.
This filter also supports the framesync options.
Turn certain luma values into transparency.
The filter accepts the following options:
Set the luma which will be used as base for transparency.
Default value is 0.
Set the range of luma values to be keyed out.
Default value is 0.
Set the range of softness. Default value is 0.
Use this to control gradual transition from zero to full transparency.
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:
set first pixel component expression
set second pixel component expression
set third pixel component expression
set fourth pixel component expression, corresponds to the alpha component
set red component expression
set green component expression
set blue component expression
alpha component expression
set Y/luminance component expression
set U/Cb component expression
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:
The input width and height.
The input value for the pixel component.
The input value, clipped to the minval-maxval range.
The maximum value for the pixel component.
The minimum value for the pixel component.
The negated value for the pixel component value, clipped to the minval-maxval range; it corresponds to the expression "maxval-clipval+minval".
The computed value in val, clipped to the minval-maxval range.
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".
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"
lutyuv=y=negval
lutyuv="u=128:v=128"
lutyuv="y=2*val"
lutrgb="g=0:b=0"
format=rgba,lutrgb=a="maxval-minval/2"
lutyuv=y=gammaval(0.5)
lutyuv=y='bitand(val, 128+64+32)'
lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'
The lut2 filter takes two input streams and outputs one
stream.
The tlut2 (time lut2) filter takes two consecutive frames
from one single stream.
This filter accepts the following parameters:
set first pixel component expression
set second pixel component expression
set third pixel component expression
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:
The input width and height.
The first input value for the pixel component.
The second input value for the pixel component.
The first input video bit depth.
The second input video bit depth.
All expressions default to "x".
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)'
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)'
lut2='if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1)))'
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:
Default value is 0.
Default value is 0.
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.
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:
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.
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:
Set the deinterlacing mode.
It accepts one of the following values:
use iterative motion estimation
like ‘slow’, but use multiple reference frames.
Default value is ‘fast’.
Set the picture field parity assumed for the input video. It must be one of the following values:
assume top field first
assume bottom field first
Default value is ‘bff’.
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.
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:
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.
Set output pixel format. Default is yuva444p.
[a0][a1][a2]mergeplanes=0x001020:yuv444p
[a0][a1]mergeplanes=0x00010210:yuva444p
format=yuva444p,mergeplanes=0x03010200:yuva444p
format=yuv420p,mergeplanes=0x000201:yuv420p
format=rgb24,mergeplanes=0x000102:yuv444p
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:
Specify the motion estimation method. Accepts one of the following values:
Exhaustive search algorithm.
Three step search algorithm.
Two dimensional logarithmic search algorithm.
New three step search algorithm.
Four step search algorithm.
Diamond search algorithm.
Hexagon-based search algorithm.
Enhanced predictive zonal search algorithm.
Uneven multi-hexagon search algorithm.
Default value is ‘esa’.
Macroblock size. Default 16.
Search parameter. Default 7.
Apply Midway Image Equalization effect using two video streams.
Midway Image Equalization adjusts a pair of images to have the same histogram, while maintaining their dynamics as much as possible. It’s useful for e.g. matching exposures from a pair of stereo cameras.
This filter has two inputs and one output, which must be of same pixel format, but may be of different sizes. The output of filter is first input adjusted with midway histogram of both inputs.
This filter accepts the following option:
Set which planes to process. Default is 15, which is all available planes.
Convert the video to specified frame rate using motion interpolation.
This filter accepts the following options:
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.
Motion interpolation mode. Following values are accepted:
Duplicate previous or next frame for interpolating new ones.
Blend source frames. Interpolated frame is mean of previous and next frames.
Motion compensated interpolation. Following options are effective when this mode is selected:
Motion compensation mode. Following values are accepted:
Overlapped block motion compensation.
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’.
Motion estimation mode. Following values are accepted:
Bidirectional motion estimation. Motion vectors are estimated for each source frame in both forward and backward directions.
Bilateral motion estimation. Motion vectors are estimated directly for interpolated frame.
Default mode is ‘bilat’.
The algorithm to be used for motion estimation. Following values are accepted:
Exhaustive search algorithm.
Three step search algorithm.
Two dimensional logarithmic search algorithm.
New three step search algorithm.
Four step search algorithm.
Diamond search algorithm.
Hexagon-based search algorithm.
Enhanced predictive zonal search algorithm.
Uneven multi-hexagon search algorithm.
Default algorithm is ‘epzs’.
Macroblock size. Default 16.
Motion estimation search parameter. Default 32.
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).
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:
Disable scene change detection.
Frame difference. Corresponding pixel values are compared and if it satisfies scd_threshold scene change is detected.
Default method is ‘fdiff’.
Scene change detection threshold. Default is 5.0.
Mix several video input streams into one video stream.
A description of the accepted options follows.
The number of inputs. If unspecified, it defaults to 2.
Specify weight of each input video stream as sequence. Each weight is separated by space.
Specify how end of stream is determined.
The duration of the longest input. (default)
The duration of the shortest input.
The duration of the first input.
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.
Set the maximum number of consecutive frames which can be dropped (if positive), or the minimum interval between dropped frames (if negative). If the value is 0, the frame is dropped disregarding the number of previous sequentially dropped frames.
Default value is 0.
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 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.
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 ‘p’x‘p’. Patches are searched in an area of ‘r’x‘r’ 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.
Set denoising strength.
Set patch size.
Same as ‘p’ but for chroma planes.
The default value is 0 and means automatic.
Set research size.
Same as ‘r’ but for chroma planes.
The default value is 0 and means automatic.
Deinterlace video using neural network edge directed interpolation.
This filter accepts the following options:
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
Set which frames to deinterlace, by default it is all.
Can be all or interlaced.
Set mode of operation.
Can be one of the following:
Use frame flags, both fields.
Use frame flags, single field.
Use top field only.
Use bottom field only.
Use both fields, top first.
Use both fields, bottom first.
Set which planes to process, by default filter process all frames.
Set size of local neighborhood around each pixel, used by the predictor neural network.
Can be one of the following:
Set the number of neurons in predictor neural network. Can be one of the following:
Controls the number of different neural network predictions that are blended
together to compute the final output value. Can be fast, default or
slow.
Set which set of weights to use in the predictor. Can be one of the following:
weights trained to minimize absolute error
weights trained to minimize squared error
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:
Default is new.
Set various debugging flags.
Force libavfilter not to use any of the specified pixel formats for the input to the next filter.
It accepts the following parameters:
A ’|’-separated list of pixel format names, such as pix_fmts=yuv420p|monow|rgb24".
noformat=pix_fmts=yuv420p,vflip
noformat=yuv420p|yuv444p|yuv410p
Add noise on video input frame.
The filter accepts the following options:
Set noise seed for specific pixel component or all pixel components in case
of all_seed. Default value is 123457.
Set noise strength for specific pixel component or all pixel components in case
all_strength. Default value is 0. Allowed range is [0, 100].
Set pixel component flags or set flags for all components if all_flags. Available values for component flags are:
averaged temporal noise (smoother)
mix random noise with a (semi)regular pattern
temporal noise (noise pattern changes between frames)
uniform noise (gaussian otherwise)
Add temporal and uniform noise to input video:
noise=alls=20:allf=t+u
Normalize RGB video (aka histogram stretching, contrast stretching). See: https://en.wikipedia.org/wiki/Normalization_(image_processing)
For each channel of each frame, the filter computes the input range and maps it linearly to the user-specified output range. The output range defaults to the full dynamic range from pure black to pure white.
Temporal smoothing can be used on the input range to reduce flickering (rapid changes in brightness) caused when small dark or bright objects enter or leave the scene. This is similar to the auto-exposure (automatic gain control) on a video camera, and, like a video camera, it may cause a period of over- or under-exposure of the video.
The R,G,B channels can be normalized independently, which may cause some color shifting, or linked together as a single channel, which prevents color shifting. Linked normalization preserves hue. Independent normalization does not, so it can be used to remove some color casts. Independent and linked normalization can be combined in any ratio.
The normalize filter accepts the following options:
Colors which define the output range. The minimum input value is mapped to the blackpt. The maximum input value is mapped to the whitept. The defaults are black and white respectively. Specifying white for blackpt and black for whitept will give color-inverted, normalized video. Shades of grey can be used to reduce the dynamic range (contrast). Specifying saturated colors here can create some interesting effects.
The number of previous frames to use for temporal smoothing. The input range of each channel is smoothed using a rolling average over the current frame and the smoothing previous frames. The default is 0 (no temporal smoothing).
Controls the ratio of independent (color shifting) channel normalization to linked (color preserving) normalization. 0.0 is fully linked, 1.0 is fully independent. Defaults to 1.0 (fully independent).
Overall strength of the filter. 1.0 is full strength. 0.0 is a rather expensive no-op. Defaults to 1.0 (full strength).
Stretch video contrast to use the full dynamic range, with no temporal smoothing; may flicker depending on the source content:
normalize=blackpt=black:whitept=white:smoothing=0
As above, but with 50 frames of temporal smoothing; flicker should be reduced, depending on the source content:
normalize=blackpt=black:whitept=white:smoothing=50
As above, but with hue-preserving linked channel normalization:
normalize=blackpt=black:whitept=white:smoothing=50:independence=0
As above, but with half strength:
normalize=blackpt=black:whitept=white:smoothing=50:independence=0:strength=0.5
Map the darkest input color to red, the brightest input color to cyan:
normalize=blackpt=red:whitept=cyan
Pass the video source unchanged to the output.
Optical Character Recognition
This filter uses Tesseract for optical character recognition.
It accepts the following options:
Set datapath to tesseract data. Default is to use whatever was set at installation.
Set language, default is "eng".
Set character whitelist.
Set character blacklist.
The filter exports recognized text as the frame metadata lavfi.ocr.text.
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:
The name of the libopencv filter to apply.
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 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 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 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.
2D Video Oscilloscope.
Useful to measure spatial impulse, step responses, chroma delays, etc.
It accepts the following parameters:
Set scope center x position.
Set scope center y position.
Set scope size, relative to frame diagonal.
Set scope tilt/rotation.
Set trace opacity.
Set trace center x position.
Set trace center y position.
Set trace width, relative to width of frame.
Set trace height, relative to height of frame.
Set which components to trace. By default it traces first three components.
Draw trace grid. By default is enabled.
Draw some statistics. By default is enabled.
Draw scope. By default is enabled.
oscilloscope=x=0.5:y=0:s=1
oscilloscope=x=0.5:y=1:s=1
oscilloscope=x=0.5:y=5/1080:s=1
oscilloscope=x=1:y=0.5:s=1:t=1
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.
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).
See framesync.
Set when the expressions for ‘x’, and ‘y’ are evaluated.
It accepts the following values:
only evaluate expressions once during the filter initialization or when a command is processed
evaluate expressions for each incoming frame
Default value is ‘frame’.
See framesync.
Set the format for the output video.
It accepts the following values:
force YUV420 output
force YUV422 output
force YUV444 output
force packed RGB output
force planar RGB output
automatically pick format
Default value is ‘yuv420’.
See framesync.
Set format of alpha of the overlaid video, it can be straight or premultiplied. Default is straight.
The ‘x’, and ‘y’ expressions can contain the following parameters.
The main input width and height.
The overlay input width and height.
The computed values for x and y. They are evaluated for each new frame.
horizontal and vertical chroma subsample values of the output format. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
the number of input frame, starting from 0
the position in the file of the input frame, NAN if unknown
The timestamp, expressed in seconds. It’s NAN if the input timestamp is unknown.
This filter also supports the framesync options.
Note that the n, pos, t variables are available only when evaluation is done per frame, and will evaluate to NAN when ‘eval’ is set to ‘init’.
Be aware that frames are taken from each input video in timestamp order, hence, if their initial timestamps differ, it is a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp, as the example for the movie filter does.
You can chain together more overlays but you should test the efficiency of such approach.
This filter supports the following commands:
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.
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
ffmpeg tool with the -filter_complex option:
ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output
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
WxH
must specify the size of the main input to the overlay filter:
color=color=red@.3:size=WxH [over]; [in][over] overlay [out]
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'
overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0
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] "
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
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]
Apply Overcomplete Wavelet denoiser.
The filter accepts the following options:
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.
Set luma strength.
Must be a double value in the range 0-1000, default is 1.0.
Set chroma strength.
Must be a double value in the range 0-1000, default is 1.0.
Add paddings to the input image, and place the original input at the provided x, y coordinates.
It accepts the following parameters:
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.
Specify the offsets to place the input image at within the padded area, with respect to the top/left border of the output image.
The x expression can reference the value set by the y expression, and vice versa.
The default value of x and y is 0.
If x or y evaluate to a negative number, they’ll be changed so the input image is centered on the padded area.
Specify the color of the padded area. For the syntax of this option, check the (ffmpeg-utils)"Color" section in the ffmpeg-utils manual.
The default value of color is "black".
Specify when to evaluate width, height, x and y expression.
It accepts the following values:
Only evaluate expressions once during the filter initialization or when a command is processed.
Evaluate expressions for each incoming frame.
Default value is ‘init’.
Pad to aspect instead to a resolution.
The value for the width, height, x, and y options are expressions containing the following constants:
The input video width and height.
These are the same as in_w and in_h.
The output width and height (the size of the padded area), as specified by the width and height expressions.
These are the same as out_w and out_h.
The x and y offsets as specified by the x and y expressions, or NAN if not yet specified.
same as iw / ih
input sample aspect ratio
input display aspect ratio, it is the same as (iw / ih) * sar
The horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
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="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"
pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"
pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"
(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"
pad="2*iw:2*ih:ow-iw:oh-ih"
Generate one palette for a whole video stream.
It accepts the following options:
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.
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.
Set the color that will be used as background for transparency.
Set statistics mode.
It accepts the following values:
Compute full frame histograms.
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.
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.
ffmpeg:
ffmpeg -i input.mkv -vf palettegen palette.png
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:
Select dithering mode. Available algorithms are:
Ordered 8x8 bayer dithering (deterministic)
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 and Steingberg dithering (error diffusion)
Frankie Sierra dithering v2 (error diffusion)
Frankie Sierra dithering v2 "Lite" (error diffusion)
Default is sierra2_4a.
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.
If set, define the zone to process
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.
Take new palette for each output frame.
Sets the alpha threshold for transparency. Alpha values above this threshold will be treated as completely opaque, and values below this threshold will be treated as completely transparent.
The option must be an integer value in the range [0,255]. Default is 128.
ffmpeg:
ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif
Correct perspective of video not recorded perpendicular to the screen.
A description of the accepted parameters follows.
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:
the width and height of video frame.
Input frame count.
Output frame count.
Set interpolation for perspective correction.
It accepts the following values:
Default value is ‘linear’.
Set interpretation of coordinate options.
It accepts the following values:
Send point in the source specified by the given coordinates to the corners of the destination.
Send the corners of the source to the point in the destination specified by the given coordinates.
Default value is ‘source’.
Set when the expressions for coordinates ‘x0,y0,...x3,y3’ are evaluated.
It accepts the following values:
only evaluate expressions once during the filter initialization or when a command is processed
evaluate expressions for each incoming frame
Default value is ‘init’.
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.
Set phase mode.
It accepts the following values:
Capture field order top-first, transfer bottom-first. Filter will delay the bottom field.
Capture field order bottom-first, transfer top-first. Filter will delay the top field.
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.
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’.
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.
Capture top-first, transfer unknown or varying. Filter selects among ‘t’ and ‘p’ using image analysis.
Capture bottom-first, transfer unknown or varying. Filter selects among ‘b’ and ‘p’ using image analysis.
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.
Both capture and transfer unknown or varying. Filter selects among ‘t’, ‘b’ and ‘p’ using image analysis only.
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.
Display sample values of color channels. Mainly useful for checking color and levels. Minimum supported resolution is 640x480.
The filters accept the following options:
Set scope X position, relative offset on X axis.
Set scope Y position, relative offset on Y axis.
Set scope width.
Set scope height.
Set window opacity. This window also holds statistics about pixel area.
Set window X position, relative offset on X axis.
Set window Y position, relative offset on Y axis.
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:
Set postprocessing subfilters string.
All subfilters share common options to determine their scope:
Honor the quality commands for this subfilter.
Do chrominance filtering, too (default).
Do luminance filtering only (no chrominance).
Do chrominance filtering only (no luminance).
These options can be appended after the subfilter name, separated by a ’|’.
Available subfilters are:
Horizontal deblocking filter
Difference factor where higher values mean more deblocking (default: 32).
Flatness threshold where lower values mean more deblocking (default: 39).
Vertical deblocking filter
Difference factor where higher values mean more deblocking (default: 32).
Flatness threshold where lower values mean more deblocking (default: 39).
Accurate horizontal deblocking filter
Difference factor where higher values mean more deblocking (default: 32).
Flatness threshold where lower values mean more deblocking (default: 39).
Accurate vertical deblocking filter
Difference factor where higher values mean more deblocking (default: 32).
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.
Experimental horizontal deblocking filter
Experimental vertical deblocking filter
Deringing filter
larger -> stronger filtering
larger -> stronger filtering
larger -> stronger filtering
Stretch luminance to 0-255.
Linear blend deinterlacing filter that deinterlaces the given block by
filtering all lines with a (1 2 1) filter.
Linear interpolating deinterlacing filter that deinterlaces the given block by linearly interpolating every second line.
Cubic interpolating deinterlacing filter deinterlaces the given block by cubically interpolating every second line.
Median deinterlacing filter that deinterlaces the given block by applying a median filter to every second line.
FFmpeg deinterlacing filter that deinterlaces the given block by filtering every
second line with a (-1 4 2 4 -1) filter.
Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given
block by filtering all lines with a (-1 2 6 2 -1) filter.
Overrides the quantizer table from the input with the constant quantizer you specify.
Quantizer to use
Default pp filter combination (hb|a,vb|a,dr|a)
Fast pp filter combination (h1|a,v1|a,dr|a)
High quality pp filter combination (ha|a|128|7,va|a,dr|a)
pp=hb/vb/dr/al
pp=de/-al
pp=default/tmpnoise|1|2|3
pp=hb|y/vb|a
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:
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).
Set thresholding mode. Available modes are:
Set hard thresholding.
Set soft thresholding (better de-ringing effect, but likely blurrier).
Set medium thresholding (good results, default).
Apply alpha premultiply effect to input video stream using first plane of second stream as alpha.
Both streams must have same dimensions and same pixel format.
The filter accepts the following option:
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
Do not require 2nd input for processing, instead use alpha plane from input stream.
Apply prewitt operator to input video stream.
The filter accepts the following option:
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
Set value which will be multiplied with filtered result.
Set value which will be added to filtered result.
Filter video using an OpenCL program.
OpenCL program source file.
Kernel name in program.
Number of inputs to the filter. Defaults to 1.
Size of output frames. Defaults to the same as the first input.
The program source file must contain a kernel function with the given name, which will be run once for each plane of the output. Each run on a plane gets enqueued as a separate 2D global NDRange with one work-item for each pixel to be generated. The global ID offset for each work-item is therefore the coordinates of a pixel in the destination image.
The kernel function needs to take the following arguments:
This image will become the output; the kernel should write all of it.
This is a counter starting from zero and increasing by one for each frame.
These are the most recent images on each input. The kernel may read from them to generate the output, but they can’t be written to.
Example programs:
__kernel void copy(__write_only image2d_t destination,
unsigned int index,
__read_only image2d_t source)
{
const sampler_t sampler = CLK_NORMALIZED_COORDS_FALSE;
int2 location = (int2)(get_global_id(0), get_global_id(1));
float4 value = read_imagef(source, sampler, location);
write_imagef(destination, location, value);
}
__kernel void rotate_image(__write_only image2d_t dst,
unsigned int index,
__read_only image2d_t src)
{
const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
CLK_FILTER_LINEAR);
float angle = (float)index / 100.0f;
float2 dst_dim = convert_float2(get_image_dim(dst));
float2 src_dim = convert_float2(get_image_dim(src));
float2 dst_cen = dst_dim / 2.0f;
float2 src_cen = src_dim / 2.0f;
int2 dst_loc = (int2)(get_global_id(0), get_global_id(1));
float2 dst_pos = convert_float2(dst_loc) - dst_cen;
float2 src_pos = {
cos(angle) * dst_pos.x - sin(angle) * dst_pos.y,
sin(angle) * dst_pos.x + cos(angle) * dst_pos.y
};
src_pos = src_pos * src_dim / dst_dim;
float2 src_loc = src_pos + src_cen;
if (src_loc.x < 0.0f || src_loc.y < 0.0f ||
src_loc.x > src_dim.x || src_loc.y > src_dim.y)
write_imagef(dst, dst_loc, 0.5f);
else
write_imagef(dst, dst_loc, read_imagef(src, sampler, src_loc));
}
__kernel void blend_images(__write_only image2d_t dst,
unsigned int index,
__read_only image2d_t src1,
__read_only image2d_t src2)
{
const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
CLK_FILTER_LINEAR);
float blend = (cos((float)index / 50.0f) + 1.0f) / 2.0f;
int2 dst_loc = (int2)(get_global_id(0), get_global_id(1));
int2 src1_loc = dst_loc * get_image_dim(src1) / get_image_dim(dst);
int2 src2_loc = dst_loc * get_image_dim(src2) / get_image_dim(dst);
float4 val1 = read_imagef(src1, sampler, src1_loc);
float4 val2 = read_imagef(src2, sampler, src2_loc);
write_imagef(dst, dst_loc, val1 * blend + val2 * (1.0f - blend));
}
Alter frame colors in video with pseudocolors.
This filter accept the following options:
set pixel first component expression
set pixel second component expression
set pixel third component expression
set pixel fourth component expression, corresponds to the alpha component
set component to use as base for altering colors
Each of them specifies the expression to use for computing the lookup table for the corresponding pixel component values.
The expressions can contain the following constants and functions:
The input width and height.
The input value for the pixel component.
The minimum allowed component value.
The maximum allowed component value.
All expressions default to "val".
pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"
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.
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.
Specifies which version of the stats file format to use. Details of each format are written below. Default value is 1.
Determines whether the max value is output to the stats log. Default value is 0. Requires stats_version >= 2. If this is set and stats_version < 2, the filter will return an error.
This filter also supports the framesync options.
The file printed if stats_file is selected, contains a sequence of key/value pairs of the form key:value for each compared couple of frames.
If a stats_version greater than 1 is specified, a header line precedes the list of per-frame-pair stats, with key value pairs following the frame format with the following parameters:
The version of the log file format. Will match stats_version.
A comma separated list of the per-frame-pair parameters included in the log.
A description of each shown per-frame-pair parameter follows:
sequential number of the input frame, starting from 1
Mean Square Error pixel-by-pixel average difference of the compared frames, averaged over all the image components.
Mean Square Error pixel-by-pixel average difference of the compared frames for the component specified by the suffix.
Peak Signal to Noise ratio of the compared frames for the component specified by the suffix.
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’.
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:
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.
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.
Set the metric plane to use. It accepts the following values:
Use luma plane.
Use chroma blue plane.
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 ...
Change video quantization parameters (QP).
The filter accepts the following option:
Set expression for quantization parameter.
The expression is evaluated through the eval API and can contain, among others, the following constants:
1 if index is not 129, 0 otherwise.
Sequential index starting from -129 to 128.
qp=2+2*sin(PI*qp)
Flush video frames from internal cache of frames into a random order. No frame is discarded. Inspired by frei0r nervous filter.
Set size in number of frames of internal cache, in range from 2 to
512. Default is 30.
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.
Read closed captioning (EIA-608) information from the top lines of a video frame.
This filter adds frame metadata for lavfi.readeia608.X.cc and
lavfi.readeia608.X.line, where X is the number of the identified line
with EIA-608 data (starting from 0). A description of each metadata value follows:
The two bytes stored as EIA-608 data (printed in hexadecimal).
The number of the line on which the EIA-608 data was identified and read.
This filter accepts the following options:
Set the line to start scanning for EIA-608 data. Default is 0.
Set the line to end scanning for EIA-608 data. Default is 29.
Set minimal acceptable amplitude change for sync codes detection.
Default is 0.2. Allowed range is [0.001 - 1].
Set the ratio of width reserved for sync code detection.
Default is 0.27. Allowed range is [0.01 - 0.7].
Set the max peaks height difference for sync code detection.
Default is 0.1. Allowed range is [0.0 - 0.5].
Set max peaks period difference for sync code detection.
Default is 0.1. Allowed range is [0.0 - 0.5].
Set the first two max start code bits differences.
Default is 0.02. Allowed range is [0.0 - 0.5].
Set the minimum ratio of bits height compared to 3rd start code bit.
Default is 0.75. Allowed range is [0.01 - 1].
Set the white color threshold. Default is 0.35. Allowed range is [0.1 - 1].
Set the black color threshold. Default is 0.15. Allowed range is [0.0 - 0.5].
Enable checking the parity bit. In the event of a parity error, the filter will output
0x00 for that character. Default is false.
ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv
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:
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.
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.
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.
--:--:--:-- 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 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.
The removegrain filter is a spatial denoiser for progressive video.
Set mode for the first plane.
Set mode for the second plane.
Set mode for the third plane.
Set mode for the fourth plane.
Range of mode is from 0 to 24. Description of each mode follows:
Leave input plane unchanged. Default.
Clips the pixel with the minimum and maximum of the 8 neighbour pixels.
Clips the pixel with the second minimum and maximum of the 8 neighbour pixels.
Clips the pixel with the third minimum and maximum of the 8 neighbour pixels.
Clips the pixel with the fourth minimum and maximum of the 8 neighbour pixels. This is equivalent to a median filter.
Line-sensitive clipping giving the minimal change.
Line-sensitive clipping, intermediate.
Line-sensitive clipping, intermediate.
Line-sensitive clipping, intermediate.
Line-sensitive clipping on a line where the neighbours pixels are the closest.
Replaces the target pixel with the closest neighbour.
[1 2 1] horizontal and vertical kernel blur.
Same as mode 11.
Bob mode, interpolates top field from the line where the neighbours pixels are the closest.
Bob mode, interpolates bottom field from the line where the neighbours pixels are the closest.
Bob mode, interpolates top field. Same as 13 but with a more complicated interpolation formula.
Bob mode, interpolates bottom field. Same as 14 but with a more complicated interpolation formula.
Clips the pixel with the minimum and maximum of respectively the maximum and minimum of each pair of opposite neighbour pixels.
Line-sensitive clipping using opposite neighbours whose greatest distance from the current pixel is minimal.
Replaces the pixel with the average of its 8 neighbours.
Averages the 9 pixels ([1 1 1] horizontal and vertical blur).
Clips pixels using the averages of opposite neighbour.
Same as mode 21 but simpler and faster.
Small edge and halo removal, but reputed useless.
Similar as 23.
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:
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.
This filter uses the repeat_field flag from the Video ES headers and hard repeats fields based on its value.
Reverse a video clip.
Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.
trim=end=5,reverse
Apply roberts cross operator to input video stream.
The filter accepts the following option:
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
Set value which will be multiplied with filtered result.
Set value which will be added to filtered result.
Rotate video by an arbitrary angle expressed in radians.
The filter accepts the following options:
A description of the optional parameters follows.
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.
Set the output width expression, default value is "iw". This expression is evaluated just once during configuration.
Set the output height expression, default value is "ih". This expression is evaluated just once during configuration.
Enable bilinear interpolation if set to 1, a value of 0 disables it. Default value is 1.
Set the color used to fill the output area not covered by the rotated image. For the general syntax of this option, check the (ffmpeg-utils)"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:
sequential number of the input frame, starting from 0. It is always NAN before the first frame is filtered.
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.
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
the input video width and height
the output width and height, that is the size of the padded area as specified by the width and height expressions
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.
rotate=PI/6
rotate=-PI/6
rotate=45*PI/180
rotate=PI/3+2*PI*t/T
rotate=A*sin(2*PI/T*t)
rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'
rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none
The filter supports the following commands:
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.
Apply Shape Adaptive Blur.
The filter accepts the following options:
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.
Set luma pre-filter radius, must be a value in the 0.1-2.0 range, default value is 1.0.
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.
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.
Set chroma pre-filter radius, must be a value in the -0.9-2.0 range.
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 (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.
The filter accepts the following options, or any of the options supported by the libswscale scaler.
See (ffmpeg-scaler)the ffmpeg-scaler manual for the complete list of scaler options.
Set the output video dimension expression. Default value is the input dimension.
If the width or w value is 0, the input width is used for the output. If the height or h value is 0, the input height is used for the output.
If one and only one of the values is -n with n >= 1, the scale filter will use a value that maintains the aspect ratio of the input image, calculated from the other specified dimension. After that it will, however, make sure that the calculated dimension is divisible by n and adjust the value if necessary.
If both values are -n with n >= 1, the behavior will be identical to both values being set to 0 as previously detailed.
See below for the list of accepted constants for use in the dimension expression.
Specify when to evaluate width and height expression. It accepts the following values:
Only evaluate expressions once during the filter initialization or when a command is processed.
Evaluate expressions for each incoming frame.
Default value is ‘init’.
Set the interlacing mode. It accepts the following values:
Force interlaced aware scaling.
Do not apply interlaced scaling.
Select interlaced aware scaling depending on whether the source frames are flagged as interlaced or not.
Default value is ‘0’.
Set libswscale scaling flags. See (ffmpeg-scaler)the ffmpeg-scaler manual for the complete list of values. If not explicitly specified the filter applies the default flags.
Set libswscale input parameters for scaling algorithms that need them. See (ffmpeg-scaler)the ffmpeg-scaler manual for the complete documentation. If not explicitly specified the filter applies empty parameters.
Set the video size. For the syntax of this option, check the (ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
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:
Choose automatically.
Format conforming to International Telecommunication Union (ITU) Recommendation BT.709.
Set color space conforming to the United States Federal Communications Commission (FCC) Code of Federal Regulations (CFR) Title 47 (2003) 73.682 (a).
Set color space conforming to:
Set color space conforming to SMPTE ST 240:1999.
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:
Choose automatically.
Set full range (0-255 in case of 8-bit luma).
Set "MPEG" range (16-235 in case of 8-bit luma).
Enable decreasing or increasing output video width or height if necessary to keep the original aspect ratio. Possible values:
Scale the video as specified and disable this feature.
The output video dimensions will automatically be decreased if needed.
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:
The input width and height
These are the same as in_w and in_h.
The output (scaled) width and height
These are the same as out_w and out_h
The same as iw / ih
input sample aspect ratio
The input display aspect ratio. Calculated from (iw / ih) * sar.
horizontal and vertical input chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
horizontal and vertical output chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
scale=w=200:h=100
This is equivalent to:
scale=200:100
or:
scale=200x100
scale=qcif
which can also be written as:
scale=size=qcif
scale=w=2*iw:h=2*ih
scale=2*in_w:2*in_h
scale=2*iw:2*ih:interl=1
scale=w=iw/2:h=ih/2
scale=3/2*iw:ow
scale=iw:1/PHI*iw scale=ih*PHI:ih
scale=w=3/2*oh:h=3/5*ih
scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"
scale=w='min(500\, iw*3/2):h=-1'
scale='trunc(ih*dar):ih',setsar=1/1
scale='trunc(ih*dar/2)*2:trunc(ih/2)*2',setsar=1/1
This filter supports the following commands:
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.
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:
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
The interpolation algorithm used for resizing. One of the following:
Nearest neighbour.
2-parameter cubic (B=1, C=0)
2-parameter cubic (B=0, C=1/2)
2-parameter cubic (B=1/2, C=3/10)
Supersampling
Scale (resize) the input video, based on a reference video.
See the scale filter for available options, scale2ref supports the same but uses the reference video instead of the main input as basis. scale2ref also supports the following additional constants for the ‘w’ and ‘h’ options:
The main input video’s width and height
The same as main_w / main_h
The main input video’s sample aspect ratio
The main input video’s display aspect ratio. Calculated from
(main_w / main_h) * main_sar.
The main input video’s horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
'scale2ref[b][a];[a][b]overlay'
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:
Select color correction method.
Available values are:
Specified adjustments are applied "as-is" (added/subtracted to original pixel component value).
Specified adjustments are relative to the original component value.
Default is absolute.
Adjustments for red pixels (pixels where the red component is the maximum)
Adjustments for yellow pixels (pixels where the blue component is the minimum)
Adjustments for green pixels (pixels where the green component is the maximum)
Adjustments for cyan pixels (pixels where the red component is the minimum)
Adjustments for blue pixels (pixels where the blue component is the maximum)
Adjustments for magenta pixels (pixels where the green component is the minimum)
Adjustments for white pixels (pixels where all components are greater than 128)
Adjustments for all pixels except pure black and pure white
Adjustments for black pixels (pixels where all components are lesser than 128)
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.
selectivecolor=greens=.5 0 -.33 0:blues=0 .27
selectivecolor=psfile=MySelectiveColorPresets/Misty.asv
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.
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:
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.
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:
These are approximated values for the mathematical constants e (Euler’s number), pi (Greek pi), and phi (the golden ratio).
The input width and height.
These are the same as w / h.
The input sample aspect ratio.
The input display aspect ratio. It is the same as (w / h) * sar.
Horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p" hsub is 2 and vsub is 1.
setdar=dar=1.77777 setdar=dar=16/9
setsar=sar=10/11
setdar=ratio=16/9:max=1000
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:
Available values are:
Keep the same field property.
Mark the frame as bottom-field-first.
Mark the frame as top-field-first.
Mark the frame as progressive.
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:
The (sequential) number of the input frame, starting from 0.
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.
The Presentation TimeStamp of the input frame, expressed as a number of seconds.
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).
The pixel format name.
The sample aspect ratio of the input frame, expressed in the form num/den.
The size of the input frame. For the syntax of this option, check the (ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
The type of interlaced mode ("P" for "progressive", "T" for top field first, "B" for bottom field first).
This is 1 if the frame is a key frame, 0 otherwise.
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’.
The Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame.
The Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, expressed in the form "[c0 c1 c2 c3]".
Displays the 256 colors palette of each frame. This filter is only relevant for pal8 pixel format frames.
It accepts the following option:
Set the size of the box used to represent one palette color entry. Default is
30 (for a 30x30 pixel box).
Reorder and/or duplicate and/or drop video frames.
It accepts the following parameters:
Set the destination indexes of input frames. This is space or ’|’ separated list of indexes that maps input frames to output frames. Number of indexes also sets maximal value that each index may have. ’-1’ index have special meaning and that is to drop frame.
The first frame has the index 0. The default is to keep the input unchanged.
ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT
ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT
Reorder and/or duplicate video planes.
It accepts the following parameters:
The index of the input plane to be used as the first output plane.
The index of the input plane to be used as the second output plane.
The index of the input plane to be used as the third output plane.
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.
ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT
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:
Display the minimal Y value contained within the input frame. Expressed in range of [0-255].
Display the Y value at the 10% percentile within the input frame. Expressed in range of [0-255].
Display the average Y value within the input frame. Expressed in range of [0-255].
Display the Y value at the 90% percentile within the input frame. Expressed in range of [0-255].
Display the maximum Y value contained within the input frame. Expressed in range of [0-255].
Display the minimal U value contained within the input frame. Expressed in range of [0-255].
Display the U value at the 10% percentile within the input frame. Expressed in range of [0-255].
Display the average U value within the input frame. Expressed in range of [0-255].
Display the U value at the 90% percentile within the input frame. Expressed in range of [0-255].
Display the maximum U value contained within the input frame. Expressed in range of [0-255].
Display the minimal V value contained within the input frame. Expressed in range of [0-255].
Display the V value at the 10% percentile within the input frame. Expressed in range of [0-255].
Display the average V value within the input frame. Expressed in range of [0-255].
Display the V value at the 90% percentile within the input frame. Expressed in range of [0-255].
Display the maximum V value contained within the input frame. Expressed in range of [0-255].
Display the minimal saturation value contained within the input frame. Expressed in range of [0-~181.02].
Display the saturation value at the 10% percentile within the input frame. Expressed in range of [0-~181.02].
Display the average saturation value within the input frame. Expressed in range of [0-~181.02].
Display the saturation value at the 90% percentile within the input frame. Expressed in range of [0-~181.02].
Display the maximum saturation value contained within the input frame. Expressed in range of [0-~181.02].
Display the median value for hue within the input frame. Expressed in range of [0-360].
Display the average value for hue within the input frame. Expressed in range of [0-360].
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].
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].
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].
Display bit depth of Y plane in current frame. Expressed in range of [0-16].
Display bit depth of U plane in current frame. Expressed in range of [0-16].
Display bit depth of V plane in current frame. Expressed in range of [0-16].
The filter accepts the following options:
‘stat’ specify an additional form of image analysis. ‘out’ output video with the specified type of pixel highlighted.
Both options accept the following values:
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.
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.
Identify pixels that fall outside of legal broadcast range.
Set the highlight color for the ‘out’ option. The default color is yellow.
ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames
ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN
ffplay example.mov -vf signalstats="out=brng:color=red"
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}
Calculates the MPEG-7 Video Signature. The filter can handle more than one input. In this case the matching between the inputs can be calculated additionally. The filter always passes through the first input. The signature of each stream can be written into a file.
It accepts the following options:
Enable or disable the matching process.
Available values are:
Disable the calculation of a matching (default).
Calculate the matching for the whole video and output whether the whole video matches or only parts.
Calculate only until a matching is found or the video ends. Should be faster in some cases.
Set the number of inputs. The option value must be a non negative integer. Default value is 1.
Set the path to which the output is written. If there is more than one input, the path must be a prototype, i.e. must contain %d or %0nd (where n is a positive integer), that will be replaced with the input number. If no filename is specified, no output will be written. This is the default.
Choose the output format.
Available values are:
Use the specified binary representation (default).
Use the specified xml representation.
Set threshold to detect one word as similar. The option value must be an integer greater than zero. The default value is 9000.
Set threshold to detect all words as similar. The option value must be an integer greater than zero. The default value is 60000.
Set threshold to detect frames as similar. The option value must be an integer greater than zero. The default value is 116.
Set the minimum length of a sequence in frames to recognize it as matching sequence. The option value must be a non negative integer value. The default value is 0.
Set the minimum relation, that matching frames to all frames must have. The option value must be a double value between 0 and 1. The default value is 0.5.
ffmpeg -i input.mkv -vf signature=filename=signature.bin -map 0:v -f null -
ffmpeg -i input1.mkv -i input2.mkv -filter_complex "[0:v][1:v] signature=nb_inputs=2:detectmode=full:format=xml:filename=signature%d.xml" -map :v -f null -
Blur the input video without impacting the outlines.
It accepts the following options:
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.
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.
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.
Set the chroma radius. The option value must be a float number in the range [0.1,5.0] that specifies the variance of the gaussian filter used to blur the image (slower if larger). Default value is ‘luma_radius’.
Set the chroma strength. The option value must be a float number in the range [-1.0,1.0] that configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included in [-1.0,0.0] will sharpen the image. Default value is ‘luma_strength’.
Set the chroma threshold used as a coefficient to determine whether a pixel should be blurred or not. The option value must be an integer in the range [-30,30]. A value of 0 will filter all the image, a value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges. Default value is ‘luma_threshold’.
If a chroma option is not explicitly set, the corresponding luma value is set.
Obtain the SSIM (Structural SImilarity Metric) between two input videos.
This filter takes in input two input videos, the first input is considered the "main" source and is passed unchanged to the output. The second input is used as a "reference" video for computing the SSIM.
Both video inputs must have the same resolution and pixel format for this filter to work correctly. Also it assumes that both inputs have the same number of frames, which are compared one by one.
The filter stores the calculated SSIM of each frame.
The description of the accepted parameters follows.
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:
sequential number of the input frame, starting from 1
SSIM of the compared frames for the component specified by the suffix.
SSIM of the compared frames for the whole frame.
Same as above but in dB representation.
This filter also supports the framesync options.
For example:
movie=ref_movie.mpg, setpts=PTS-STARTPTS [main]; [main][ref] ssim="stats_file=stats.log" [out]
On this example the input file being processed is compared with the reference file ‘ref_movie.mpg’. The SSIM of each individual frame is stored in ‘stats.log’.
Another example with both psnr and ssim at same time:
ffmpeg -i main.mpg -i ref.mpg -lavfi "ssim;[0:v][1:v]psnr" -f null -
Convert between different stereoscopic image formats.
The filters accept the following options:
Set stereoscopic image format of input.
Available values for input image formats are:
side by side parallel (left eye left, right eye right)
side by side crosseye (right eye left, left eye right)
side by side parallel with half width resolution (left eye left, right eye right)
side by side crosseye with half width resolution (right eye left, left eye right)
above-below (left eye above, right eye below)
above-below (right eye above, left eye below)
above-below with half height resolution (left eye above, right eye below)
above-below with half height resolution (right eye above, left eye below)
alternating frames (left eye first, right eye second)
alternating frames (right eye first, left eye second)
interleaved rows (left eye has top row, right eye starts on next row)
interleaved rows (right eye has top row, left eye starts on next row)
interleaved columns, left eye first
interleaved columns, right eye first
Default value is ‘sbsl’.
Set stereoscopic image format of output.
side by side parallel (left eye left, right eye right)
side by side crosseye (right eye left, left eye right)
side by side parallel with half width resolution (left eye left, right eye right)
side by side crosseye with half width resolution (right eye left, left eye right)
above-below (left eye above, right eye below)
above-below (right eye above, left eye below)
above-below with half height resolution (left eye above, right eye below)
above-below with half height resolution (right eye above, left eye below)
alternating frames (left eye first, right eye second)
alternating frames (right eye first, left eye second)
interleaved rows (left eye has top row, right eye starts on next row)
interleaved rows (right eye has top row, left eye starts on next row)
anaglyph red/blue gray (red filter on left eye, blue filter on right eye)
anaglyph red/green gray (red filter on left eye, green filter on right eye)
anaglyph red/cyan gray (red filter on left eye, cyan filter on right eye)
anaglyph red/cyan half colored (red filter on left eye, cyan filter on right eye)
anaglyph red/cyan color (red filter on left eye, cyan filter on right eye)
anaglyph red/cyan color optimized with the least squares projection of dubois (red filter on left eye, cyan filter on right eye)
anaglyph green/magenta gray (green filter on left eye, magenta filter on right eye)
anaglyph green/magenta half colored (green filter on left eye, magenta filter on right eye)
anaglyph green/magenta colored (green filter on left eye, magenta filter on right eye)
anaglyph green/magenta color optimized with the least squares projection of dubois (green filter on left eye, magenta filter on right eye)
anaglyph yellow/blue gray (yellow filter on left eye, blue filter on right eye)
anaglyph yellow/blue half colored (yellow filter on left eye, blue filter on right eye)
anaglyph yellow/blue colored (yellow filter on left eye, blue filter on right eye)
anaglyph yellow/blue color optimized with the least squares projection of dubois (yellow filter on left eye, blue filter on right eye)
mono output (left eye only)
mono output (right eye only)
checkerboard, left eye first
checkerboard, right eye first
interleaved columns, left eye first
interleaved columns, right eye first
HDMI frame pack
Default value is ‘arcd’.
stereo3d=sbsl:aybd
stereo3d=abl:sbsr
Select video or audio streams.
The filter accepts the following options:
Set number of inputs. Default is 2.
Set input indexes to remap to outputs.
The streamselect and astreamselect filter supports the following
commands:
Set input indexes to remap to outputs.
sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0
asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0
Apply sobel operator to input video stream.
The filter accepts the following option:
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
Set value which will be multiplied with filtered result.
Set value which will be added to filtered result.
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:
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.
Force a constant quantization parameter. If not set, the filter will use the QP from the video stream (if available).
Set thresholding mode. Available modes are:
Set hard thresholding (default).
Set soft thresholding (better de-ringing effect, but likely blurrier).
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).
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:
Set the filename of the subtitle file to read. It must be specified.
Specify the size of the original video, the video for which the ASS file was composed. For the syntax of this option, check the (ffmpeg-utils)"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.
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.
Process alpha channel, by default alpha channel is untouched.
Set subtitles input character encoding. subtitles filter only. Only
useful if not UTF-8.
Set subtitles stream index. subtitles filter only.
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'
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.
Swap two rectangular objects in video.
This filter accepts the following options:
Set object width.
Set object height.
Set 1st rect x coordinate.
Set 1st rect y coordinate.
Set 2nd rect x coordinate.
Set 2nd rect y coordinate.
All expressions are evaluated once for each frame.
The all options are expressions containing the following constants:
The input width and height.
same as w / h
input sample aspect ratio
input display aspect ratio, it is the same as (w / h) * sar
The number of the input frame, starting from 0.
The timestamp expressed in seconds. It’s NAN if the input timestamp is unknown.
the position in the file of the input frame, NAN if unknown
Swap U & V plane.
Apply telecine process to the video.
This filter accepts the following options:
top field first
bottom field first
The default value is top.
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
Apply threshold effect to video stream.
This filter needs four video streams to perform thresholding. First stream is stream we are filtering. Second stream is holding threshold values, third stream is holding min values, and last, fourth stream is holding max values.
The filter accepts the following option:
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
For example if first stream pixel’s component value is less then threshold value of pixel component from 2nd threshold stream, third stream value will picked, otherwise fourth stream pixel component value will be picked.
Using color source filter one can perform various types of thresholding:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=black -f lavfi -i color=white -lavfi threshold output.avi
ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -f lavfi -i color=black -lavfi threshold output.avi
ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=gray -lavfi threshold output.avi
ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -i 320x240.avi -lavfi threshold output.avi
ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=white -lavfi threshold output.avi
Select the most representative frame in a given sequence of consecutive frames.
The filter accepts the following options:
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.
thumbnail=50
ffmpeg:
ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png
Tile several successive frames together.
The filter accepts the following options:
Set the grid size (i.e. the number of lines and columns). For the syntax of this option, check the (ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
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.
Set the outer border margin in pixels.
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.
Specify the color of the unused area. For the syntax of this option, check the (ffmpeg-utils)"Color" section in the ffmpeg-utils manual. The default value of color is "black".
Set the number of frames to overlap when tiling several successive frames together.
The value must be between 0 and nb_frames - 1.
Set the number of frames to initially be empty before displaying first output frame.
This controls how soon will one get first output frame.
The value must be between 0 and nb_frames - 1.
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.
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
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:
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:
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
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
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
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 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 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
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
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.
Specify flags influencing the filter process.
Available value for flags is:
Enable linear vertical low-pass filtering in the filter. Vertical low-pass filtering is required when creating an interlaced destination from a progressive source which contains high-frequency vertical detail. Filtering will reduce interlace ’twitter’ and Moire patterning.
Enable complex vertical low-pass filtering. This will slightly less reduce interlace ’twitter’ and Moire patterning but better retain detail and subjective sharpness impression.
Vertical low-pass filtering can only be enabled for ‘mode’ interleave_top and interleave_bottom.
Tone map colors from different dynamic ranges.
This filter expects data in single precision floating point, as it needs to operate on (and can output) out-of-range values. Another filter, such as zscale, is needed to convert the resulting frame to a usable format.
The tonemapping algorithms implemented only work on linear light, so input data should be linearized beforehand (and possibly correctly tagged).
ffmpeg -i INPUT -vf zscale=transfer=linear,tonemap=clip,zscale=transfer=bt709,format=yuv420p OUTPUT
The filter accepts the following options.
Set the tone map algorithm to use.
Possible values are:
Do not apply any tone map, only desaturate overbright pixels.
Hard-clip any out-of-range values. Use it for perfect color accuracy for in-range values, while distorting out-of-range values.
Stretch the entire reference gamut to a linear multiple of the display.
Fit a logarithmic transfer between the tone curves.
Preserve overall image brightness with a simple curve, using nonlinear contrast, which results in flattening details and degrading color accuracy.
Preserve both dark and bright details better than reinhard, at the cost of slightly darkening everything. Use it when detail preservation is more important than color and brightness accuracy.
Smoothly map out-of-range values, while retaining contrast and colors for in-range material as much as possible. Use it when color accuracy is more important than detail preservation.
Default is none.
Tune the tone mapping algorithm.
This affects the following algorithms:
Ignored.
Specifies the scale factor to use while stretching. Default to 1.0.
Specifies the exponent of the function. Default to 1.8.
Specify an extra linear coefficient to multiply into the signal before clipping. Default to 1.0.
Specify the local contrast coefficient at the display peak. Default to 0.5, which means that in-gamut values will be about half as bright as when clipping.
Ignored.
Specify the transition point from linear to mobius transform. Every value below this point is guaranteed to be mapped 1:1. The higher the value, the more accurate the result will be, at the cost of losing bright details. Default to 0.3, which due to the steep initial slope still preserves in-range colors fairly accurately.
Apply desaturation for highlights that exceed this level of brightness. The higher the parameter, the more color information will be preserved. This setting helps prevent unnaturally blown-out colors for super-highlights, by (smoothly) turning into white instead. This makes images feel more natural, at the cost of reducing information about out-of-range colors.
The default of 2.0 is somewhat conservative and will mostly just apply to skies or directly sunlit surfaces. A setting of 0.0 disables this option.
This option works only if the input frame has a supported color tag.
Override signal/nominal/reference peak with this value. Useful when the embedded peak information in display metadata is not reliable or when tone mapping from a lower range to a higher range.
Transpose rows with columns in the input video and optionally flip it.
It accepts the following parameters:
Specify the transposition direction.
Can assume the following values:
Rotate by 90 degrees counterclockwise and vertically flip (default), that is:
L.R L.l . . -> . . l.r R.r
Rotate by 90 degrees clockwise, that is:
L.R l.L . . -> . . l.r r.R
Rotate by 90 degrees counterclockwise, that is:
L.R R.r . . -> . . l.r L.l
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.
Do not apply the transposition if the input geometry matches the one specified by the specified value. It accepts the following values:
Always apply transposition.
Preserve portrait geometry (when height >= width).
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 the input so that the output contains one continuous subpart of the input.
It accepts the following parameters:
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.
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.
This is the same as start, except this option sets the start timestamp in timebase units instead of seconds.
This is the same as end, except this option sets the end timestamp in timebase units instead of seconds.
The maximum duration of the output in seconds.
The number of the first frame that should be passed to the output.
The number of the first frame that should be dropped.
‘start’, ‘end’, and ‘duration’ are expressed as time duration specifications; see (ffmpeg-utils)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:
ffmpeg -i INPUT -vf trim=60:120
ffmpeg -i INPUT -vf trim=duration=1
Apply alpha unpremultiply effect to input video stream using first plane of second stream as alpha.
Both streams must have same dimensions and same pixel format.
The filter accepts the following option:
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
If the format has 1 or 2 components, then luma is bit 0. If the format has 3 or 4 components: for RGB formats bit 0 is green, bit 1 is blue and bit 2 is red; for YUV formats bit 0 is luma, bit 1 is chroma-U and bit 2 is chroma-V. If present, the alpha channel is always the last bit.
Do not require 2nd input for processing, instead use alpha plane from input stream.
Sharpen or blur the input video.
It accepts the following parameters:
Set the luma matrix horizontal size. It must be an odd integer between 3 and 23. The default value is 5.
Set the luma matrix vertical size. It must be an odd integer between 3 and 23. The default value is 5.
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.
Set the chroma matrix horizontal size. It must be an odd integer between 3 and 23. The default value is 5.
Set the chroma matrix vertical size. It must be an odd integer between 3 and 23. The default value is 5.
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.
All parameters are optional and default to the equivalent of the string ’5:5:1.0:5:5:0.0’.
unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5
unsharp=7:7:-2:7:7:-2
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:
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.
Force a constant quantization parameter. If not set, the filter will use the QP from the video stream (if available).
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:
The filtering strength. The higher, the more filtered the video will be. Hard thresholding can use a higher threshold than soft thresholding before the video looks overfiltered. Default value is 2.
The filtering method the filter will use.
It accepts the following values:
All values under the threshold will be zeroed.
All values under the threshold will be zeroed. All values above will be reduced by the threshold.
Scales or nullifies coefficients - intermediary between (more) soft and (less) hard thresholding.
Default is garrote.
Number of times, the wavelet will decompose the picture. Picture can’t be decomposed beyond a particular point (typically, 8 for a 640x480 frame - as 2^9 = 512 > 480). Valid values are integers between 1 and 32. Default value is 6.
Partial of full denoising (limited coefficients shrinking), from 0 to 100. Default value is 85.
A list of the planes to process. By default all planes are processed.
Display 2 color component values in the two dimensional graph (which is called a vectorscope).
This filter accepts the following options:
Set vectorscope mode.
It accepts the following values:
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.
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.
Actual color components values present in video frame are displayed on graph.
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.
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.
Gray values are displayed on graph. Similar to color but with 3rd color
component picked from radial gradient.
Set which color component will be represented on X-axis. Default is 1.
Set which color component will be represented on Y-axis. Default is 2.
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.
No envelope, this is default.
Instant envelope, even darkest single pixel will be clearly highlighted.
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 and instant envelope combined together.
Set what kind of graticule to draw.
Set graticule opacity.
Set graticule flags.
Draw graticule for white point.
Draw graticule for black point.
Draw color points short names.
Set background opacity.
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.
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.
Set what kind of colorspace to use when drawing graticule.
Default is auto.
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:
Set the path to the file used to write the transforms information. Default value is ‘transforms.trf’.
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.
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.
Set stepsize of the search process. The region around minimum is scanned with 1 pixel resolution. Default value is 6.
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.
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 fields and transforms in the resulting frames. It accepts an integer in the range 0-2. Default value is 0, which disables any visualization.
vidstabdetect
vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"
vidstabdetect=show=1
ffmpeg:
ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi
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.
Set path to the file used to read the transforms. Default value is ‘transforms.trf’.
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.
Set the camera path optimization algorithm.
Accepted values are:
gaussian kernel low-pass filter on camera motion (default)
averaging on transformations
Set maximal number of pixels to translate frames. Default value is -1, meaning no limit.
Set maximal angle in radians (degree*PI/180) to rotate frames. Default value is -1, meaning no limit.
Specify how to deal with borders that may be visible due to movement compensation.
Available values are:
keep image information from previous frame (default)
fill the border black
Invert transforms if set to 1. Default value is 0.
Consider transforms as relative to previous frame if set to 1, absolute if set to 0. Default value is 0.
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).
Set optimal zooming to avoid borders.
Accepted values are:
disabled
optimal static zoom value is determined (only very strong movements will lead to visible borders) (default)
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.
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.
Specify type of interpolation.
Available values are:
no interpolation
linear only horizontal
linear in both directions (default)
cubic in both directions (slow)
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.
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.
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.
vidstabtransform=zoom=5:input="mytransforms.trf"
vidstabtransform=smoothing=30
Flip the input video vertically.
For example, to vertically flip a video with ffmpeg:
ffmpeg -i in.avi -vf "vflip" out.avi
Detect variable frame rate video.
This filter tries to detect if the input is variable or constant frame rate.
At end it will output number of frames detected as having variable delta pts, and ones with constant delta pts. If there was frames with variable delta, than it will also show min and max delta encountered.
Make or reverse a natural vignetting effect.
The filter accepts the following options:
Set lens angle expression as a number of radians.
The value is clipped in the [0,PI/2] range.
Default value: "PI/5"
Set center coordinates expressions. Respectively "w/2" and "h/2"
by default.
Set forward/backward mode.
Available modes are:
The larger the distance from the central point, the darker the image becomes.
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’.
Set evaluation mode for the expressions (‘angle’, ‘x0’, ‘y0’).
It accepts the following values:
Evaluate expressions only once during the filter initialization.
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’.
Set dithering to reduce the circular banding effects. Default is 1
(enabled).
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.
The ‘alpha’, ‘x0’ and ‘y0’ expressions can contain the following parameters.
input width and height
the number of input frame, starting from 0
the PTS (Presentation TimeStamp) time of the filtered video frame, expressed in TB units, NAN if undefined
frame rate of the input video, NAN if the input frame rate is unknown
the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined
time base of the input video
vignette=PI/4
vignette='PI/4+random(1)*PI/50':eval=frame
Obtain the average vmaf motion score of a video. It is one of the component filters of VMAF.
The obtained average motion score is printed through the logging system.
In the below example the input file ‘ref.mpg’ is being processed and score is computed.
ffmpeg -i ref.mpg -lavfi vmafmotion -f null -
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:
Set number of input streams. Default is 2.
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.
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:
Set the interlacing filter coefficients. Accepts one of the following values:
Simple filter coefficient set.
More-complex filter coefficient set.
Default value is ‘complex’.
Specify which frames to deinterlace. Accept one of the following values:
Deinterlace all frames,
Only deinterlace frames marked as interlaced.
Default value is ‘all’.
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:
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.
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].
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).
Set display mode. It accepts the following values:
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.
Display separate graph for the color components side by side in
row mode or one below the other in column mode.
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.
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.
No envelope, this is default.
Instant envelope, minimum and maximum values presented in graph will be easily
visible even with small step value.
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 and instant envelope combined together.
No filtering, this is default.
Luma and chroma combined together.
Similar as above, but shows difference between blue and red chroma.
Similar as above, but use different colors.
Displays only chroma.
Displays actual color value on waveform.
Similar as above, but with luma showing frequency of chroma values.
Set which graticule to display.
Do not display graticule.
Display green graticule showing legal broadcast ranges.
Display orange graticule showing legal broadcast ranges.
Set graticule opacity.
Set graticule flags.
Draw numbers above lines. By default enabled.
Draw dots instead of lines.
Set scale used for displaying graticule.
Default is digital.
Set background opacity.
The weave takes a field-based video input and join
each two sequential fields into single frame, producing a new double
height clip with half the frame rate and half the frame count.
The doubleweave works same as weave but without
halving frame rate and frame count.
It accepts the following option:
Set first field. Available values are:
Set the frame as top-field-first.
Set the frame as bottom-field-first.
separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave
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:
Set the scaling dimension: 2 for 2xBR, 3 for
3xBR and 4 for 4xBR.
Default is 3.
Deinterlace the input video ("yadif" means "yet another deinterlacing filter").
It accepts the following parameters:
The interlacing mode to adopt. It accepts one of the following values:
Output one frame for each frame.
Output one frame for each field.
Like send_frame, but it skips the spatial interlacing check.
Like send_field, but it skips the spatial interlacing check.
The default value is send_frame.
The picture field parity assumed for the input interlaced video. It accepts one of the following values:
Assume the top field is first.
Assume the bottom field is first.
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.
Specify which frames to deinterlace. Accept one of the following values:
Deinterlace all frames.
Only deinterlace frames marked as interlaced.
The default value is all.
Apply Zoom & Pan effect.
This filter accepts the following options:
Set the zoom expression. Default is 1.
Set the x and y expression. Default is 0.
Set the duration expression in number of frames. This sets for how many number of frames effect will last for single input image.
Set the output image size, default is ’hd720’.
Set the output frame rate, default is ’25’.
Each expression can contain the following constants:
Input width.
Input height.
Output width.
Output height.
Input frame count.
Output frame count.
Last calculated ’x’ and ’y’ position from ’x’ and ’y’ expression for current input frame.
’x’ and ’y’ of last output frame of previous input frame or 0 when there was not yet such frame (first input frame).
Last calculated zoom from ’z’ expression for current input frame.
Last calculated zoom of last output frame of previous input frame.
Number of output frames for current input frame. Calculated from ’d’ expression for each input frame.
number of output frames created for previous input frame
Rational number: input width / input height
sample aspect ratio
display aspect ratio
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
zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'
zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'
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.
The filter accepts the following options.
Set the output video dimension expression. Default value is the input dimension.
If the width or w value is 0, the input width is used for the output. If the height or h value is 0, the input height is used for the output.
If one and only one of the values is -n with n >= 1, the zscale filter will use a value that maintains the aspect ratio of the input image, calculated from the other specified dimension. After that it will, however, make sure that the calculated dimension is divisible by n and adjust the value if necessary.
If both values are -n with n >= 1, the behavior will be identical to both values being set to 0 as previously detailed.
See below for the list of accepted constants for use in the dimension expression.
Set the video size. For the syntax of this option, check the (ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Set the dither type.
Possible values are:
Default is none.
Set the resize filter type.
Possible values are:
Default is bilinear.
Set the color range.
Possible values are:
Default is same as input.
Set the color primaries.
Possible values are:
Default is same as input.
Set the transfer characteristics.
Possible values are:
Default is same as input.
Set the colorspace matrix.
Possible value are:
Default is same as input.
Set the input color range.
Possible values are:
Default is same as input.
Set the input color primaries.
Possible values are:
Default is same as input.
Set the input transfer characteristics.
Possible values are:
Default is same as input.
Set the input colorspace matrix.
Possible value are:
Set the output chroma location.
Possible values are:
Set the input chroma location.
Possible values are:
Set the nominal peak luminance.
The values of the ‘w’ and ‘h’ options are expressions containing the following constants:
The input width and height
These are the same as in_w and in_h.
The output (scaled) width and height
These are the same as out_w and out_h
The same as iw / ih
input sample aspect ratio
The input display aspect ratio. Calculated from (iw / ih) * sar.
horizontal and vertical input chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
horizontal and vertical output chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
Below is a description of the currently available video sources.
Buffer video frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular through the interface defined in ‘libavfilter/vsrc_buffer.h’.
It accepts the following parameters:
Specify the size (width and height) of the buffered video frames. For the syntax of this option, check the (ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
The input video width.
The input video height.
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.
Specify the timebase assumed by the timestamps of the buffered frames.
Specify the frame rate expected for the video stream.
The sample (pixel) aspect ratio of the input video.
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.
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]
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:
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.
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.
Set the video rate, that is the number of frames generated per second. Default is 25.
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.
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.
Set the cellular automaton rule, it is a number ranging from 0 to 255. Default value is 110.
Set the size of the output video. For the syntax of this option, check the (ffmpeg-utils)"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).
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.
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.
If set to 1, stitch the left and right row edges together. This is the default behavior, for disabling set the value to 0.
cellauto=f=pattern:s=200x400
cellauto=ratio=2/3:s=200x200
cellauto=p=@:s=100x400:full=0:rule=18
cellauto=p='@@ @ @@':s=100x400:full=0:rule=18
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 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
Specify the size of the sourced video. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
The default value is 320x240.
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".
Set the sample aspect ratio of the sourced video.
Set the duration of the sourced video. See (ffmpeg-utils)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.
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.
Generate a Mandelbrot set fractal, and progressively zoom towards the point specified with start_x and start_y.
This source accepts the following options:
Set the terminal pts value. Default value is 400.
Set the terminal scale value. Must be a floating point value. Default value is 0.3.
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:
Set black mode.
Show time until convergence.
Set color based on point closest to the origin of the iterations.
Set period mode.
Default value is mincol.
Set the bailout value. Default value is 10.0.
Set the maximum of iterations performed by the rendering algorithm. Default value is 7189.
Set outer coloring mode. It shall assume one of following values:
Set iteration cound mode.
set normalized iteration count mode.
Default value is normalized_iteration_count.
Set frame rate, expressed as number of frames per second. Default value is "25".
Set frame size. For the syntax of this option, check the (ffmpeg-utils)"Video size" section in the ffmpeg-utils manual. Default value is "640x480".
Set the initial scale value. Default value is 3.0.
Set the initial x position. Must be a floating point value between -100 and 100. Default value is -0.743643887037158704752191506114774.
Set the initial y position. Must be a floating point value between -100 and 100. Default value is -0.131825904205311970493132056385139.
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:
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".
Set the duration of the sourced video. See (ffmpeg-utils)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.
Set the number or the name of the test to perform. Supported tests are:
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.
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:
The size of the video to generate. For the syntax of this option, check the (ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
The framerate of the generated video. It may be a string of the form num/den or a frame rate abbreviation.
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.
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
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:
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.
Set the video rate, that is the number of frames generated per second. Default is 25.
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.
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.
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.
Set the size of the output video. For the syntax of this option, check the (ffmpeg-utils)"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).
If set to 1, stitch the left and right grid edges together, and the top and bottom edges also. Defaults to 1.
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.
Set the color of living (or new born) cells.
Set the color of dead cells. If ‘mold’ is set, this is the first color used to represent a dead cell.
Set mold color, for definitely dead and moldy cells.
For the syntax of these 3 color options, check the (ffmpeg-utils)"Color" section in the ffmpeg-utils manual.
life=f=pattern:s=300x300
life=ratio=2/3:s=200x200
life=rule=S14/B34
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
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:
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.
Specify the color of the source, only available in the color
source. For the syntax of this option, check the
(ffmpeg-utils)"Color" section in the ffmpeg-utils manual.
Specify the size of the sourced video. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
The default value is 320x240.
This option is not available with the allrgb, allyuv, and
haldclutsrc filters.
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".
Set the duration of the sourced video. See (ffmpeg-utils)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.
Set the sample aspect ratio of the sourced video.
Specify the alpha (opacity) of the background, only available in the
testsrc2 source. The value must be between 0 (fully transparent) and
255 (fully opaque, the default).
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.
testsrc=duration=5.3:size=qcif:rate=10
color=c=red@0.2:s=qcif:r=10
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
The color source supports the following commands:
Set the color of the created image. Accepts the same syntax of the corresponding ‘color’ option.
Generate video using an OpenCL program.
OpenCL program source file.
Kernel name in program.
Size of frames to generate. This must be set.
Pixel format to use for the generated frames. This must be set.
Number of frames generated every second. Default value is ’25’.
For details of how the program loading works, see the program_opencl filter.
Example programs:
__kernel void ramp(__write_only image2d_t dst,
unsigned int index)
{
int2 loc = (int2)(get_global_id(0), get_global_id(1));
float4 val;
val.xy = val.zw = convert_float2(loc) / convert_float2(get_image_dim(dst));
write_imagef(dst, loc, val);
}
__kernel void sierpinski_carpet(__write_only image2d_t dst,
unsigned int index)
{
int2 loc = (int2)(get_global_id(0), get_global_id(1));
float4 value = 0.0f;
int x = loc.x + index;
int y = loc.y + index;
while (x > 0 || y > 0) {
if (x % 3 == 1 && y % 3 == 1) {
value = 1.0f;
break;
}
x /= 3;
y /= 3;
}
write_imagef(dst, loc, value);
}
Below is a description of the currently available video sinks.
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.
Null video sink: do absolutely nothing with the input video. It is mainly useful as a template and for use in analysis / debugging tools.
Below is a description of the currently available multimedia filters.
Convert input audio to a video output, displaying the audio bit scope.
The filter accepts the following options:
Set frame rate, expressed as number of frames per second. Default value is "25".
Specify the video size for the output. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default value is 1024x256.
Specify list of colors separated by space or by ’|’ which will be used to draw channels. Unrecognized or missing colors will be replaced by white color.
Convert input audio to a video output, displaying the volume histogram.
The filter accepts the following options:
Specify how histogram is calculated.
It accepts the following values:
Use single histogram for all channels.
Use separate histogram for each channel.
Default is single.
Set frame rate, expressed as number of frames per second. Default value is "25".
Specify the video size for the output. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default value is hd720.
Set display scale.
It accepts the following values:
logarithmic
square root
cubic root
linear
reverse logarithmic
Default is log.
Set amplitude scale.
It accepts the following values:
logarithmic
linear
Default is log.
Set how much frames to accumulate in histogram. Defauls is 1. Setting this to -1 accumulates all frames.
Set histogram ratio of window height.
Set sonogram sliding.
It accepts the following values:
replace old rows with new ones.
scroll from top to bottom.
Default is replace.
Convert input audio to a video output, displaying the audio phase.
The filter accepts the following options:
Set the output frame rate. Default value is 25.
Set the video size for the output. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default value is 800x400.
Specify the red, green, blue contrast. Default values are 2,
7 and 1.
Allowed range is [0, 255].
Set color which will be used for drawing median phase. If color is
none which is default, no median phase value will be drawn.
Enable video output. Default is enabled.
The filter also exports the frame metadata lavfi.aphasemeter.phase which
represents mean phase of current audio frame. Value is in range [-1, 1].
The -1 means left and right channels are completely out of phase and
1 means channels are in phase.
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:
Set the vectorscope mode.
Available values are:
Lissajous rotated by 45 degrees.
Same as above but not rotated.
Shape resembling half of circle.
Default value is ‘lissajous’.
Set the video size for the output. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default value is 400x400.
Set the output frame rate. Default value is 25.
Specify the red, green, blue and alpha contrast. Default values are 40,
160, 80 and 255.
Allowed range is [0, 255].
Specify the red, green, blue and alpha fade. Default values are 15,
10, 5 and 5.
Allowed range is [0, 255].
Set the zoom factor. Default value is 1. Allowed range is [0, 10].
Values lower than 1 will auto adjust zoom factor to maximal possible value.
Set the vectorscope drawing mode.
Available values are:
Draw dot for each sample.
Draw line between previous and current sample.
Default value is ‘dot’.
Specify amplitude scale of audio samples.
Available values are:
Linear.
Square root.
Cubic root.
Logarithmic.
Swap left channel axis with right channel axis.
Mirror axis.
No mirror.
Mirror only x axis.
Mirror only y axis.
Mirror both axis.
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]'
Benchmark part of a filtergraph.
The filter accepts the following options:
Start or stop a timer.
Available values are:
Get the current time, set it as frame metadata (using the key
lavfi.bench.start_time), and forward the frame to the next filter.
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.
bench=start,selectivecolor=reds=-.2 .12 -.49,bench=stop
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:
Set the number of segments. Default is 2.
Set the number of output video streams, that is also the number of video streams in each segment. Default is 1.
Set the number of output audio streams, that is also the number of audio streams in each segment. Default is 0.
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.
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
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.
This filter supports the following commands:
Close the current segment and step to the next one
Draw a graph using input video or audio metadata.
It accepts the following parameters:
Set 1st frame metadata key from which metadata values will be used to draw a graph.
Set 1st foreground color expression.
Set 2nd frame metadata key from which metadata values will be used to draw a graph.
Set 2nd foreground color expression.
Set 3rd frame metadata key from which metadata values will be used to draw a graph.
Set 3rd foreground color expression.
Set 4th frame metadata key from which metadata values will be used to draw a graph.
Set 4th foreground color expression.
Set minimal value of metadata value.
Set maximal value of metadata value.
Set graph background color. Default is white.
Set graph mode.
Available values for mode is:
Default is line.
Set slide mode.
Available values for slide is:
Draw new frame when right border is reached.
Replace old columns with new ones.
Scroll from right to left.
Scroll from left to right.
Draw single picture.
Default is frame.
Set size of graph video. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
The default value is 900x256.
The foreground color expressions can use the following variables:
Minimal value of metadata value.
Maximal value of metadata value.
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
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:
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.
Set the video size. This option is for video only. For the syntax of this
option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default and minimum resolution is 640x480.
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.
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.
Force the frame logging level.
Available values are:
information logging level
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.
Set peak mode(s).
Available modes can be cumulated (the option is a flag type). Possible
values are:
Disable any peak mode (default).
Enable sample-peak mode.
Simple peak mode looking for the higher sample value. It logs a message
for sample-peak (identified by SPK).
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.
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.
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.
ffplay, with a EBU scale meter +18:
ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]"
ffmpeg:
ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null -
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:
Set the number of different inputs, it is 2 by default.
ffmpeg:
ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi
select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave
Manipulate frame metadata.
This filter accepts the following options:
Set mode of operation of the filter.
Can be one of the following:
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 new metadata key and value. If key is already available
do nothing.
Modify value of already present key.
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 key and its value if metadata was found. If key is not set print all
metadata values available in frame.
Set key used with all modes. Must be set for all modes except print and delete.
Set metadata value which will be used. This option is mandatory for
modify and add mode.
Which function to use when comparing metadata value and value.
Can be one of following:
Values are interpreted as strings, returns true if metadata value is same as value.
Values are interpreted as strings, returns true if metadata value starts with
the value option string.
Values are interpreted as floats, returns true if metadata value is less than value.
Values are interpreted as floats, returns true if value is equal with metadata value.
Values are interpreted as floats, returns true if metadata value is greater than value.
Values are interpreted as floats, returns true if expression from option expr
evaluates to true.
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:
Float representation of value from metadata key.
Float representation of value as supplied by user in value option.
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.
lavfi.signalstats.YDIF with values
between 0 and 1.
signalstats,metadata=print:key=lavfi.signalstats.YDIF:value=0:function=expr:expr='between(VALUE1,0,1)'
silencedetect,ametadata=mode=print:file=metadata.txt
metadata=mode=print:file='pipe\:4'
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:
Select the permissions mode.
It accepts the following values:
Do nothing. This is the default.
Set all the output frames read-only.
Set all the output frames directly writable.
Make the frame read-only if writable, and writable if read-only.
Set each output frame read-only or writable randomly.
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.
Slow down filtering to match real time approximately.
These filters will pause the filtering for a variable amount of time to
match the output rate with the input timestamps.
They are similar to the ‘re’ option to ffmpeg.
They accept the following options:
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 frames to pass in output.
This filter accepts the following options:
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.
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:
The (sequential) number of the filtered frame, starting from 0.
The (sequential) number of the selected frame, starting from 0.
The sequential number of the last selected frame. It’s NAN if undefined.
The timebase of the input timestamps.
The PTS (Presentation TimeStamp) of the filtered video frame, expressed in TB units. It’s NAN if undefined.
The PTS of the filtered video frame, expressed in seconds. It’s NAN if undefined.
The PTS of the previously filtered video frame. It’s NAN if undefined.
The PTS of the last previously filtered video frame. It’s NAN if undefined.
The PTS of the last previously selected video frame, expressed in seconds. It’s NAN if undefined.
The PTS of the first video frame in the video. It’s NAN if undefined.
The time of the first video frame in the video. It’s NAN if undefined.
The type of the filtered frame. It can assume one of the following values:
The frame interlace type. It can assume one of the following values:
The frame is progressive (not interlaced).
The frame is top-field-first.
The frame is bottom-field-first.
the number of selected samples before the current frame
the number of samples in the current frame
the input sample rate
This is 1 if the filtered frame is a key-frame, 0 otherwise.
the position in the file of the filtered frame, -1 if the information is not available (e.g. for synthetic video)
value between 0 and 1 to indicate a new scene; a low value reflects a low probability for the current frame to introduce a new scene, while a higher value means the current frame is more likely to be one (see the example below)
The 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".
select
The example above is the same as:
select=1
select=0
select='eq(pict_type\,I)'
select='not(mod(n\,100))'
select=between(t\,10\,20)
select=between(t\,10\,20)*eq(pict_type\,I)
select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'
aselect='gt(samples_n\,100)'
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.
select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h
ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi
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:
Set the commands to be read and sent to the other filters.
Set the filename of the commands to be read and sent to the other filters.
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:
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.
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]
asendcmd=c='4.0 atempo tempo 1.5',atempo
asendcmd=c='4.0 atempo@my tempo 1.5',atempo@my
# 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
Change the PTS (presentation timestamp) of the input frames.
setpts works on video frames, asetpts on audio frames.
This filter accepts the following options:
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, only defined for constant frame-rate video
The presentation timestamp in input
The count of the input frame for video or the number of consumed samples, not including the current frame for audio, starting from 0.
The number of consumed samples, not including the current frame (only audio)
The number of samples in the current frame (only audio)
The audio sample rate.
The PTS of the first frame.
the time in seconds of the first frame
State whether the current frame is interlaced.
the time in seconds of the current frame
original position in the file of the frame, or undefined if undefined for the current frame
The previous input PTS.
previous input time in seconds
The previous output PTS.
previous output time in seconds
The wallclock (RTC) time in microseconds. This is deprecated, use time(0) instead.
The wallclock (RTC) time at the start of the movie in microseconds.
The timebase of the input timestamps.
setpts=PTS-STARTPTS
setpts=0.5*PTS
setpts=2.0*PTS
setpts=N/(25*TB)
setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'
setpts=PTS+10/TB
setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'
asetpts=N/SR/TB
Force color range for the output video frame.
The setrange filter marks the color range property 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.
The filter accepts the following options:
Available values are:
Keep the same color range property.
Set the color range as unspecified.
Set the color range as limited.
Set the color range as full.
Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase configuration.
It accepts the following parameters:
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".
settb=expr=1/25
settb=expr=0.1
settb=1+0.001
settb=2*intb
settb=AVTB
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:
Specify the video size for the output. It must be even. For the syntax of this option,
check the (ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default value is 1920x1080.
Set the output frame rate. Default value is 25.
Set the bargraph height. It must be even. Default value is -1 which
computes the bargraph height automatically.
Set the axis height. It must be even. Default value is -1 which computes
the axis height automatically.
Set the sonogram height. It must be even. Default value is -1 which
computes the sonogram height automatically.
Set the fullhd resolution. This option is deprecated, use size, s
instead. Default value is 1.
Specify the sonogram volume expression. It can contain variables:
the bar_v evaluated expression
the frequency where it is evaluated
the value of timeclamp option
and functions:
A-weighting of equal loudness
B-weighting of equal loudness
C-weighting of equal loudness.
Default value is 16.
Specify the bargraph volume expression. It can contain variables:
the sono_v evaluated expression
the frequency where it is evaluated
the value of timeclamp option
and functions:
A-weighting of equal loudness
B-weighting of equal loudness
C-weighting of equal loudness.
Default value is sono_v.
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].
Specify the bargraph gamma. Default value is 1. Acceptable range is
[1, 7].
Specify the bargraph transparency level. Lower value makes the bargraph sharper.
Default value is 1. Acceptable range is [0, 1].
Specify the transform timeclamp. At low frequency, there is trade-off between
accuracy in time domain and frequency domain. If timeclamp is lower,
event in time domain is represented more accurately (such as fast bass drum),
otherwise event in frequency domain is represented more accurately
(such as bass guitar). Acceptable range is [0.002, 1]. Default value is 0.17.
Set attack time in seconds. The default is 0 (disabled). Otherwise, it
limits future samples by applying asymmetric windowing in time domain, useful
when low latency is required. Accepted range is [0, 1].
Specify the transform base frequency. Default value is 20.01523126408007475,
which is frequency 50 cents below E0. Acceptable range is [10, 100000].
Specify the transform end frequency. Default value is 20495.59681441799654,
which is frequency 50 cents above D#10. Acceptable range is [10, 100000].
This option is deprecated and ignored.
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:
the frequency where it is evaluated
the value of timeclamp option.
Default value is 384*tc/(384+tc*f).
Specify the transform count for every video frame. Default value is 6.
Acceptable range is [1, 30].
Specify the transform count for every single pixel. Default value is 0,
which makes it computed automatically. Acceptable range is [0, 10].
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.
Specify fontconfig pattern. This has lower priority than fontfile. The : in the pattern may be replaced by | to avoid unnecessary escaping.
Specify font color expression. This is arithmetic expression that should return integer value 0xRRGGBB. It can contain variables:
the frequency where it is evaluated
the value of timeclamp option
and functions:
midi number of frequency f, some midi numbers: E0(16), C1(24), C2(36), A4(69)
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)).
Specify image file to draw the axis. This option override fontfile and fontcolor option.
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.
Set colorspace. The accepted values are:
Unspecified (default)
BT.709
FCC
BT.470BG or BT.601-6 625
SMPTE-170M or BT.601-6 525
SMPTE-240M
BT.2020 with non-constant luminance
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.
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]'
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]'
sono_h=0
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]'
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]'
bar_v=10:sono_v=bar_v*a_weighting(f)
bar_g=2:sono_g=2
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)))'
fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))':fontfile=myfont.ttf
font='Courier New,Monospace,mono|bold'
axisfile=myaxis.png:basefreq=40:endfreq=10000
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:
Specify size of video. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default is 1024x512.
Set display mode. This set how each frequency bin will be represented.
It accepts the following values:
Default is bar.
Set amplitude scale.
It accepts the following values:
Linear scale.
Square root scale.
Cubic root scale.
Logarithmic scale.
Default is log.
Set frequency scale.
It accepts the following values:
Linear scale.
Logarithmic scale.
Reverse logarithmic scale.
Default is lin.
Set window size.
It accepts the following values:
Default is w2048
Set windowing function.
It accepts the following values:
Default is hanning.
Set window overlap. In range [0, 1]. Default is 1,
which means optimal overlap for selected window function will be picked.
Set time averaging. Setting this to 0 will display current maximal peaks.
Default is 1, which means time averaging is disabled.
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.
Set channel display mode.
It accepts the following values:
Default is combined.
Set minimum amplitude used in log amplitude scaler.
Convert input audio to a video output, representing the audio frequency spectrum.
The filter accepts the following options:
Specify the video size for the output. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default value is 640x512.
Specify how the spectrum should slide along the window.
It accepts the following values:
the samples start again on the left when they reach the right
the samples scroll from right to left
frames are only produced when the samples reach the right
the samples scroll from left to right
Default value is replace.
Specify display mode.
It accepts the following values:
all channels are displayed in the same row
all channels are displayed in separate rows
Default value is ‘combined’.
Specify display color mode.
It accepts the following values:
each channel is displayed in a separate color
each channel is displayed using the same color scheme
each channel is displayed using the rainbow color scheme
each channel is displayed using the moreland color scheme
each channel is displayed using the nebulae color scheme
each channel is displayed using the fire color scheme
each channel is displayed using the fiery color scheme
each channel is displayed using the fruit color scheme
each channel is displayed using the cool color scheme
Default value is ‘channel’.
Specify scale used for calculating intensity color values.
It accepts the following values:
linear
square root, default
cubic root
logarithmic
4th root
5th root
Default value is ‘sqrt’.
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.
Set window function.
It accepts the following values:
Default value is hann.
Set orientation of time vs frequency axis. Can be vertical or
horizontal. Default is vertical.
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.
Set scale gain for calculating intensity color values.
Default value is 1.
Set which data to display. Can be magnitude, default or phase.
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.
showspectrum=s=1280x480:scale=log
ffplay:
ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'
Convert input audio to a single video frame, representing the audio frequency spectrum.
The filter accepts the following options:
Specify the video size for the output. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default value is 4096x2048.
Specify display mode.
It accepts the following values:
all channels are displayed in the same row
all channels are displayed in separate rows
Default value is ‘combined’.
Specify display color mode.
It accepts the following values:
each channel is displayed in a separate color
each channel is displayed using the same color scheme
each channel is displayed using the rainbow color scheme
each channel is displayed using the moreland color scheme
each channel is displayed using the nebulae color scheme
each channel is displayed using the fire color scheme
each channel is displayed using the fiery color scheme
each channel is displayed using the fruit color scheme
each channel is displayed using the cool color scheme
Default value is ‘intensity’.
Specify scale used for calculating intensity color values.
It accepts the following values:
linear
square root, default
cubic root
logarithmic
4th root
5th root
Default value is ‘log’.
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.
Set window function.
It accepts the following values:
Default value is hann.
Set orientation of time vs frequency axis. Can be vertical or
horizontal. Default is vertical.
Set scale gain for calculating intensity color values.
Default value is 1.
Draw time and frequency axes and legends. Default is enabled.
Set color rotation, must be in [-1.0, 1.0] range.
Default value is 0.
ffmpeg:
ffmpeg -i audio.flac -lavfi showspectrumpic=s=1024x1024 spectrogram.png
Convert input audio volume to a video output.
The filter accepts the following options:
Set video rate.
Set border width, allowed range is [0, 5]. Default is 1.
Set channel width, allowed range is [80, 8192]. Default is 400.
Set channel height, allowed range is [1, 900]. Default is 20.
Set fade, allowed range is [0, 1]. Default is 0.95.
Set volume color expression.
The expression can use the following variables:
Current max volume of channel in dB.
Current peak.
Current channel number, starting from 0.
If set, displays channel names. Default is enabled.
If set, displays volume values. Default is enabled.
Set orientation, can be horizontal: h or vertical: v,
default is h.
Set step size, allowed range is [0, 5]. Default is 0, which means step is disabled.
Set background opacity, allowed range is [0, 1]. Default is 0.
Set metering mode, can be peak: p or rms: r,
default is p.
Set display scale, can be linear: lin or log: log,
default is lin.
In second.
If set to > 0., display a line for the max level
in the previous seconds.
default is disabled: 0.
The color of the max line. Use when dm option is set to > 0.
default is: orange
Convert input audio to a video output, representing the samples waves.
The filter accepts the following options:
Specify the video size for the output. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default value is 600x240.
Set display mode.
Available values are:
Draw a point for each sample.
Draw a vertical line for each sample.
Draw a point for each sample and a line between them.
Draw a centered vertical line for each sample.
Default value is point.
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.
Set the (approximate) output frame rate. This is done by setting the option n. Default value is "25".
Set if channels should be drawn separately or overlap. Default value is 0.
Set colors separated by ’|’ which are going to be used for drawing of each channel.
Set amplitude scale.
Available values are:
Linear.
Logarithmic.
Square root.
Cubic root.
Default is linear.
Set the draw mode. This is mostly useful to set for high n.
Available values are:
Scale pixel values for each drawn sample.
Draw every sample directly.
Default value is scale.
amovie=a.mp3,asplit[out0],showwaves[out1]
aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1]
Convert input audio to a single video frame, representing the samples waves.
The filter accepts the following options:
Specify the video size for the output. For the syntax of this option, check the
(ffmpeg-utils)"Video size" section in the ffmpeg-utils manual.
Default value is 600x240.
Set if channels should be drawn separately or overlap. Default value is 0.
Set colors separated by ’|’ which are going to be used for drawing of each channel.
Set amplitude scale.
Available values are:
Linear.
Logarithmic.
Square root.
Cubic root.
Default is linear.
ffmpeg:
ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png
Delete frame side data, or select frames based on it.
This filter accepts the following options:
Set mode of operation of the filter.
Can be one of the following:
Select every frame with side data of type.
Delete side data of type. If type is not set, delete all side
data in the frame.
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.
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:
Specify sample rate of output audio, the sample rate of audio from which spectrum was generated may differ.
Set number of channels represented in input video spectrums.
Set scale which was used when generating magnitude input spectrum.
Can be lin or log. Default is log.
Set slide which was used when generating inputs spectrums.
Can be replace, scroll, fullframe or rscroll.
Default is fullframe.
Set window function used for resynthesis.
Set window overlap. In range [0, 1]. Default is 1,
which means optimal overlap for selected window function will be picked.
Set orientation of input videos. Can be vertical or horizontal.
Default is vertical.
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 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.
[in] split [out0][out1]
[in] asplit=3 [out0][out1][out2]
[in] split [splitout1][splitout2]; [splitout1] crop=100:100:0:0 [cropout]; [splitout2] pad=200:200:100:100 [padout];
ffmpeg:
ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT
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. Both are capable to send messages to any filter type.
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’ (or the abbreviation "‘b’") option.
Default value of this option is ‘tcp://localhost:5555’. You may
want to alter this value to your needs, but do not forget to escape any
’:’ signs (see filtergraph escaping).
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. The default filter instance name uses the pattern ‘Parsed_<filter_name>_<index>’, but you can override this by using the ‘filter_name@id’ syntax (see Filtergraph syntax).
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.
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.
In this example the last overlay filter has an instance name. All other
filters will have default instance names.
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@my=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
To change the position of the right side:
echo overlay@my x 150 | tools/zmqsend
Below is a description of the currently available multimedia sources.
This is the same as movie source, except it selects an audio stream by default.
Read audio and/or video stream(s) from a movie container.
It accepts the following parameters:
The name of the resource to read (not necessarily a file; it can also be a device or a stream accessed through some protocol).
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.
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".
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 (ffmpeg)"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".
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.
Specifies how many times to read the stream in sequence. If the value is 0, the stream will be looped infinitely. Default value is "1".
Note that when the movie is looped the source timestamps are not changed, so it will generate non monotonically increasing timestamps.
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 -------+
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]
movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over]; [in] setpts=PTS-STARTPTS [main]; [main][over] overlay=16:16 [out]
movie=dvd.vob:s=v:0+#0x81 [video] [audio]
Both movie and amovie support the following commands:
Perform seek using "av_seek_frame". The syntax is: seek stream_index|timestamp|flags
Get movie duration in AV_TIME_BASE units.
ffmpeg ffplay, ffprobe, ffmpeg-utils, ffmpeg-scaler, ffmpeg-resampler, ffmpeg-codecs, ffmpeg-bitstream-filters, ffmpeg-formats, ffmpeg-devices, ffmpeg-protocols, ffmpeg-filters
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.