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

ffmpeg - ffmpeg video converter

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.

  • To set the video bitrate of the output file to 64 kbit/s:
    ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi
    
  • To force the frame rate of the output file to 24 fps:
    ffmpeg -i input.avi -r 24 output.avi
    
  • To force the frame rate of the input file (valid for raw formats only) to 1 fps and the frame rate of the output file to 24 fps:
    ffmpeg -r 1 -i input.m2v -r 24 output.avi
    

The format option may be needed for raw input files.

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

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

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

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

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

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

Complex filtergraphs

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

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

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

The -lavfi option is equivalent to -filter_complex.

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

Stream copy 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.

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

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

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

Automatic stream selection

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

It will select that stream based upon the following criteria:

  • for video, it is the stream with the highest resolution,
  • for audio, it is the stream with the most channels,
  • for subtitles, it is the first subtitle stream found but there's a caveat. The output format's default subtitle encoder can be either text-based or image-based, and only a subtitle stream of the same type will be chosen.

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

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

Manual stream selection

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

Complex filtergraphs

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

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

Stream handling

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

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

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

They assume the following three input files.

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

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

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

Example: automatic stream selection

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

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

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

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

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

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

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

Example: automatic subtitles selection

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

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

Example: unlabeled filtergraph outputs

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

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

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

Example: labeled filtergraph outputs

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

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

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

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

The command should be modified as follows,

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

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

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

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

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

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

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. If stream_index is used as an additional stream specifier (see below), then it selects stream number stream_index from the matching streams. Stream numbering is based on the order of the streams as detected by libavformat except when a program ID is also specified. In this case it is based on the ordering of the streams in the program.
stream_type is one of following: 'v' or 'V' for video, 'a' for audio, 's' for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video streams, 'V' only matches video streams which are not attached pictures, video thumbnails or cover arts. If additional_stream_specifier is used, then it matches streams which both have this type and match the additional_stream_specifier. Otherwise, it matches all streams of the specified type.
Matches streams which are in the program with the id program_id. If additional_stream_specifier is used, then it matches streams which both are part of the program and match the additional_stream_specifier.
#stream_id or i:stream_id
Match the stream by stream id (e.g. PID in MPEG-TS container).
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 named filter_name. Use the -filters option to get a list of all filters.
Print detailed information about the bitstream filter named bitstream_filter_name. Use the -bsfs option to get a list of all bitstream filters.
Print detailed information about the protocol named protocol_name. Use the -protocols option to get a list of all protocols.
Show version.
Show the build configuration, one option per line.
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 can be forced setting the environment variable AV_LOG_FORCE_COLOR.

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

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

The following options are recognized:

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

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

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:

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

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

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

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

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 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 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 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 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.

Rescale input timestamps. scale should be a floating point number.
Set the recording timestamp in the container.

date must be a date specification, see 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 -c copy -disposition:a:1 default out.mkv

To make the second subtitle stream the default stream and remove the default disposition from the first subtitle stream:

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

To add an embedded cover/thumbnail:

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

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

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

The parameters set for each target are as follows.

VCD

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

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

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

SVCD

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

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

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

DVD

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

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

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

DV

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

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

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

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

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

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

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

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".
Set period at which encoding progress/statistics are updated. Default is 0.5 seconds.
Send program-friendly progress information to url.

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

The update period is set using "-stats_period".

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.

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 maximum frame rate (Hz value, fraction or abbreviation).

Clamps output frame rate when output framerate is auto-set and is higher than this value. Useful in batch processing or when input framerate is wrongly detected as very high. It cannot be set together with "-r". It is ignored during streamcopy.

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.

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

As an output option, disables video recording i.e. automatic selection or mapping of any video stream. 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.

Automatically rotate the video according to file metadata. Enabled by default, use -noautorotate to disable it.
Automatically scale the video according to the resolution of first frame. Enabled by default, use -noautoscale to disable it. When autoscale is disabled, all output frames of filter graph might not be in the same resolution and may be inadequate for some encoder/muxer. Therefore, it is not recommended to disable it unless you really know what you are doing. Disable autoscale at your own risk.

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.
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 by use of a filter such as "yadif" or "bwdif", but deinterlacing introduces losses.
-psnr
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 can take arguments of the following form:
If the argument consists of timestamps, ffmpeg will round the specified times to the nearest output timestamp as per the encoder time base and force a keyframe at the first frame having timestamp equal or greater than the computed timestamp. Note that if the encoder time base is too coarse, then the keyframes may be forced on frames with timestamps lower than the specified time. The default encoder time base is the inverse of the output framerate but may be set otherwise via "-enc_time_base".

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
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.

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))
If the argument is "source", ffmpeg will force a key frame if the current frame being encoded is marked as a key frame in its source.

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.
If device is an integer, it selects the device by its index in a system-dependent list of devices. If device is any other string, it selects the first device with a name containing that string as a substring.

The following options are recognized:

If set to 1, enables the validation layer, if installed.
If set to 1, images allocated by the hwcontext will be linear and locally mappable.
A plus separated list of additional instance extensions to enable.
A plus separated list of additional device extensions to enable.

Examples:

Choose the second device on the system.
Choose the first device with a name containing the string RADV.
Choose the first device and enable the Wayland and XCB instance extensions.
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.
As an input option, blocks all audio streams of a file from being filtered or being automatically selected or mapped for any output. See "-discard" option to disable streams individually.

As an output option, disables audio recording i.e. automatic selection or mapping of any audio stream. 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".
As an input option, blocks all subtitle streams of a file from being filtered or being automatically selected or mapped for any output. See "-discard" option to disable streams individually.

As an output option, disables subtitle recording i.e. automatic selection or mapping of any subtitle stream. 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 real, system and user 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 real, system and user time used in various steps (audio/video encode/decode).
Exit after ffmpeg has been running for duration seconds in CPU user time.
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).
Video sync method. For compatibility reasons old values can be specified as numbers. Newly added values will have to be specified as strings always.
0, passthrough
Each frame is passed with its timestamp from the demuxer to the muxer.
1, cfr
Frames will be duplicated and dropped to achieve exactly the requested constant frame rate.
2, vfr
Frames are passed through with their timestamp or dropped so as to prevent 2 frames from having the same timestamp.
As passthrough but destroys all timestamps, making the muxer generate fresh timestamps based on frame-rate.
-1, auto
Chooses between 1 and 2 depending on muxer capabilities. This is the default method.

Note that the timestamps may be further modified by the muxer, after this. For example, in the case that the format option avoid_negative_ts is enabled.

With -map you can select from which stream the timestamps should be taken. You can leave either video or audio unchanged and sync the remaining stream(s) to the unchanged one.

Frame drop threshold, which specifies how much behind video frames can be before they are dropped. In frame rate units, so 1.0 is one frame. The default is -1.1. One possible usecase is to avoid framedrops in case of noisy timestamps or to increase frame drop precision in case of exact timestamps.
-async samples_per_second
Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps, the parameter is the maximum samples per second by which the audio is changed. -async 1 is a special case where only the start of the audio stream is corrected without any later correction.

Note that the timestamps may be further modified by the muxer, after this. For example, in the case that the format option avoid_negative_ts is enabled.

This option has been deprecated. Use the "aresample" audio filter instead.

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 (add/drop) and soft (squeeze/stretch) compensation. "-async" must be set to a positive value.
-apad parameters (output,per-stream)
Pad the output audio stream(s). This is the same as applying "-af apad". Argument is a string of filter parameters composed the same as with the "apad" filter. "-shortest" must be set for this output for the option to take effect.
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:
1
Use the demuxer timebase.

The time base is copied to the output encoder from the corresponding input demuxer. This is sometimes required to avoid non monotonically increasing timestamps when copying video streams with variable frame rate.

0
Use the decoder timebase.

The time base is copied to the output encoder from the corresponding input decoder.

-1
Try to make the choice automatically, in order to generate a sane output.

Default value is -1.

Set the encoder timebase. timebase is a floating point number, and can assume one of the following values:
0
Assign a default value according to the media type.

For video - use 1/framerate, for audio - use 1/samplerate.

-1
Use the input stream timebase when possible.

If an input stream is not available, the default timebase will be used.

>0
Use the provided number as the timebase.

This field can be provided as a ratio of two integers (e.g. 1:24, 1:48000) or as a floating point number (e.g. 0.04166, 2.0833e-5)

Default value is 0.

Enable bitexact mode for (de)muxer and (de/en)coder
Finish encoding when the shortest input stream ends.
Timestamp discontinuity delta threshold.
Timestamp error delta threshold. This threshold use to discard crazy/damaged timestamps and the default is 30 hours which is arbitrarily picked and quite conservative.
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

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)

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.
-lavfi filtergraph (global)
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; setting this value can force ffmpeg to use a separate input thread and read packets as soon as they arrive. By default ffmpeg only do this if multiple inputs are specified.
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 from streams. Any input stream can be fully discarded, using value "all" whereas selective discarding of frames from a stream occurs at the demuxer and is not supported by all demuxers.
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.
No packets were passed to the muxer in some of the output streams.
Set fraction of decoding frame failures across all inputs which when crossed ffmpeg will return exit code 69. Crossing this threshold does not terminate processing. Range is a floating-point number between 0 to 1. Default is 2/3.
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.

This is a minimum threshold until which the muxing queue size is not taken into account. Defaults to 50 megabytes per stream, and is based on the overall size of packets passed to the muxer.
Enable automatically inserting format conversion filters in all filter graphs, including those defined by -vf, -af, -filter_complex and -lavfi. If filter format negotiation requires a conversion, the initialization of the filters will fail. Conversions can still be performed by inserting the relevant conversion filter (scale, aresample) in the graph. On by default, to explicitly disable it you need to specify "-noauto_conversion_filters".

A preset file contains a sequence of option=value pairs, one for each line, specifying a sequence of options which would be awkward to specify on the command line. Lines starting with the hash ('#') character are ignored and are used to provide comments. Check the presets directory in the FFmpeg source tree for examples.

There are two types of preset files: ffpreset and avpreset files.

ffpreset files

ffpreset files are specified with the "vpre", "apre", "spre", and "fpre" options. The "fpre" option takes the filename of the preset instead of a preset name as input and can be used for any kind of codec. For the "vpre", "apre", and "spre" options, the options specified in a preset file are applied to the currently selected codec of the same type as the preset option.

The argument passed to the "vpre", "apre", and "spre" preset options identifies the preset file to use according to the following rules:

First ffmpeg searches for a file named arg.ffpreset in the directories $FFMPEG_DATADIR (if set), and $HOME/.ffmpeg, and in the datadir defined at configuration time (usually PREFIX/share/ffmpeg) or in a ffpresets folder along the executable on win32, in that order. For example, if the argument is "libvpx-1080p", it will search for the file libvpx-1080p.ffpreset.

If no such file is found, then ffmpeg will search for a file named codec_name-arg.ffpreset in the above-mentioned directories, where codec_name is the name of the codec to which the preset file options will be applied. For example, if you select the video codec with "-vcodec libvpx" and use "-vpre 1080p", then it will search for the file libvpx-1080p.ffpreset.

avpreset files

avpreset files are specified with the "pre" option. They work similar to ffpreset files, but they only allow encoder- specific options. Therefore, an option=value pair specifying an encoder cannot be used.

When the "pre" option is specified, ffmpeg will look for files with the suffix .avpreset in the directories $AVCONV_DATADIR (if set), and $HOME/.avconv, and in the datadir defined at configuration time (usually PREFIX/share/ffmpeg), in that order.

First ffmpeg searches for a file named codec_name-arg.avpreset in the above-mentioned directories, where codec_name is the name of the codec to which the preset file options will be applied. For example, if you select the video codec with "-vcodec libvpx" and use "-pre 1080p", then it will search for the file libvpx-1080p.avpreset.

If no such file is found, then ffmpeg will search for a file named arg.avpreset in the same directories.

If you specify the input format and device then ffmpeg can grab video and audio directly.

ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg

Or with an ALSA audio source (mono input, card id 1) instead of OSS:

ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg

Note that you must activate the right video source and channel before launching ffmpeg with any TV viewer such as http://linux.bytesex.org/xawtv/ by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer.

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:

  • You can use YUV files as input:
    ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
    

    It will use the files:

    /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
    /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
    

    The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the -s option if ffmpeg cannot guess it.

  • You can input from a raw YUV420P file:
    ffmpeg -i /tmp/test.yuv /tmp/out.avi
    

    test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed by the U and V planes at half vertical and horizontal resolution.

  • You can output to a raw YUV420P file:
    ffmpeg -i mydivx.avi hugefile.yuv
    
  • You can set several input files and output files:
    ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
    

    Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg.

  • You can also do audio and video conversions at the same time:
    ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
    

    Converts a.wav to MPEG audio at 22050 Hz sample rate.

  • You can encode to several formats at the same time and define a mapping from input stream to output streams:
    ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2
    

    Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map file:index' specifies which input stream is used for each output stream, in the order of the definition of output streams.

  • You can transcode decrypted VOBs:
    ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi
    

    This is a typical DVD ripping example; the input is a VOB file, the output an AVI file with MPEG-4 video and MP3 audio. Note that in this command we use B-frames so the MPEG-4 stream is DivX5 compatible, and GOP size is 300 which means one intra frame every 10 seconds for 29.97fps input video. Furthermore, the audio stream is MP3-encoded so you need to enable LAME support by passing "--enable-libmp3lame" to configure. The mapping is particularly useful for DVD transcoding to get the desired audio language.

    NOTE: To see the supported input formats, use "ffmpeg -demuxers".

  • You can extract images from a video, or create a video from many images:

    For extracting images from a video:

    ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg
    

    This will extract one video frame per second from the video and will output them in files named foo-001.jpeg, foo-002.jpeg, etc. Images will be rescaled to fit the new WxH values.

    If you want to extract just a limited number of frames, you can use the above command in combination with the "-frames:v" or "-t" option, or in combination with -ss to start extracting from a certain point in time.

    For creating a video from many images:

    ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
    

    The syntax "foo-%03d.jpeg" specifies to use a decimal number composed of three digits padded with zeroes to express the sequence number. It is the same syntax supported by the C printf function, but only formats accepting a normal integer are suitable.

    When importing an image sequence, -i also supports expanding shell-like wildcard patterns (globbing) internally, by selecting the image2-specific "-pattern_type glob" option.

    For example, for creating a video from filenames matching the glob pattern "foo-*.jpeg":

    ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
    
  • You can put many streams of the same type in the output:
    ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
    

    The resulting output file test12.nut will contain the first four streams from the input files in reverse order.

  • To force CBR video output:
    ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v
    
  • The four options lmin, lmax, mblmin and mblmax use 'lambda' units, but you may use the QP2LAMBDA constant to easily convert from 'q' units:
    ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext
    

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:

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

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

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

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

Examples

  • Escape the string "Crime d'Amour" containing the "'" special character:
    Crime d\'Amour
    
  • The string above contains a quote, so the "'" needs to be escaped when quoting it:
    'Crime d'\''Amour'
    
  • Include leading or trailing whitespaces using quoting:
    '  this string starts and ends with whitespaces  '
    
  • Escaping and quoting can be mixed together:
    ' The string '\'string\'' is a string '
    
  • To include a literal \ you can use either escaping or quoting:
    'c:\foo' can be written as c:\\foo
    

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|ms|us]

S expresses the number of seconds, with the optional decimal part m. The optional literal suffixes s, ms or us indicate to interpret the value as seconds, milliseconds or microseconds, respectively.

In both expressions, the optional - indicates negative duration.

Examples

The following examples are all valid time duration:

55
55 seconds
0.2
0.2 seconds
200ms
200 milliseconds, that's 0.2s
200000us
200000 microseconds, that's 0.2s
12:03:45
12 hours, 03 minutes and 45 seconds
23.189
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
4cif
704x576
16cif
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
2k
2048x1080
2kflat
1998x1080
2kscope
2048x858
4k
4096x2160
4kflat
3996x2160
4kscope
4096x1716
640x360
240x160
400x240
432x240
480x320
960x540
2kdci
2048x1080
4kdci
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
2.1
FL+FR+LFE
3.0
FL+FR+FC
3.0(back)
FL+FR+BC
4.0
FL+FR+FC+BC
FL+FR+BL+BR
FL+FR+SL+SR
3.1
FL+FR+FC+LFE
5.0
FL+FR+FC+BL+BR
5.0(side)
FL+FR+FC+SL+SR
4.1
FL+FR+FC+LFE+BC
5.1
FL+FR+FC+LFE+BL+BR
5.1(side)
FL+FR+FC+LFE+SL+SR
6.0
FL+FR+FC+BC+SL+SR
6.0(front)
FL+FR+FLC+FRC+SL+SR
FL+FR+FC+BL+BR+BC
6.1
FL+FR+FC+LFE+BC+SL+SR
6.1
FL+FR+FC+LFE+BL+BR+BC
6.1(front)
FL+FR+LFE+FLC+FRC+SL+SR
7.0
FL+FR+FC+BL+BR+SL+SR
7.0(front)
FL+FR+FC+FLC+FRC+SL+SR
7.1
FL+FR+FC+LFE+BL+BR+SL+SR
7.1(wide)
FL+FR+FC+LFE+BL+BR+FLC+FRC
7.1(wide-side)
FL+FR+FC+LFE+FLC+FRC+SL+SR
FL+FR+FC+BL+BR+BC+SL+SR
FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR
DL+DR

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

  • the name of a standard channel layout (e.g. mono, stereo, 4.0, quad, 5.0, etc.)
  • the name of a single channel (e.g. FL, FR, FC, LFE, etc.)
  • a number of channels, in decimal, followed by 'c', yielding the default channel layout for that number of channels (see the function "av_get_default_channel_layout"). Note that not all channel counts have a default layout.
  • a number of channels, in decimal, followed by 'C', yielding an unknown channel layout with the specified number of channels. Note that not all channel layout specification strings support unknown channel layouts.
  • a channel layout mask, in hexadecimal starting with "0x" (see the "AV_CH_*" macros in libavutil/channel_layout.h.

Before libavutil version 53 the trailing character "c" to specify a number of channels was optional, but now it is required, while a channel layout mask can also be specified as a decimal number (if and only if not followed by "c" or "C").

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

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 sign of x.
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.
loop
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.
psnr
Set error[?] variables during encoding.
Input bitstream might be randomly truncated.
Don't output frames whose parameters differ from first decoded frame in stream. Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.
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.
Apply interlaced motion estimation.
Use closed gop.
Output even potentially corrupted frames.
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 strategy to choose between I/P/B-frames.
Set RTP payload size in bytes.
Workaround not auto detected encoder bugs.

Possible values:

Xvid interlacing bug (autodetected if fourcc==XVIX)
(autodetected if fourcc==UMP4)
padding bug (autodetected)
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
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
buffer
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.
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).
Set QP factor between P and I frames.
Set QP offset between P and I frames.
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)
deblock
use strong deblock filter for damaged MBs
favor predicting from the previous frame instead of the current
Set prediction method.

Possible values:

median
Set sample aspect ratio.
Set sample aspect ratio. Alias to aspect.
Print specific debug info.

Possible values:

picture info
rate control
macroblock (MB) type
qp
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
psnr
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
psnr
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
psnr
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
psnr
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.
(1024, INT_MAX)
full motion estimation(slowest)
(768, 1024]
umh motion estimation
(512, 768]
hex motion estimation
(256, 512]
l2s diamond motion estimation
[2,256]
var diamond motion estimation
(-1, 2)
small diamond motion estimation
-1
funny diamond motion estimation
(INT_MIN, -1)
sab diamond 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
psnr
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).
Possible values:
variable length coder / huffman coder
arithmetic coder
raw (no encoding)
run-length 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 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.
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.
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.
Do not skip samples and export skip information as frame side data.
Do not reset ASS ReadOrder field on flush.
Possible values:
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.
Export encoder Producer Reference Time into packet side-data (see "AV_PKT_DATA_PRFT") for codecs that support it.
Export video encoding parameters through frame side data (see "AV_FRAME_DATA_VIDEO_ENC_PARAMS") for codecs that support it. At present, those are H.264 and VP9.
Export film grain parameters through frame side data (see "AV_FRAME_DATA_FILM_GRAIN_PARAMS"). Supported at present by AV1 decoders.
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 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.
Set encoder codec profile. Default value is unknown. Encoder specific profiles are documented in the relevant encoder documentation.
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
psnr
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 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 except I frames.
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.
Adjust sensitivity of b_frame_strategy 1.
Set GOP timecode frame start number, in non drop frame format.
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
colorspace integer (decoding/encoding,video)
Possible values:
RGB
BT.709
FCC
BT.470 BG
SMPTE 170 M
SMPTE 240 M
YCOCG
BT.2020 NCL
BT.2020 CL
SMPTE 2085
Chroma-derived NCL
Chroma-derived CL
ICtCp
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 indentation:
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.

AOMedia Video 1 (AV1) decoder.

Options

Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0.

Raw video decoder.

This decoder decodes rawvideo streams.

Options

Specify the assumed field type of the input video.
-1
the video is assumed to be progressive (default)
0
bottom-field-first is assumed
1
top-field-first is assumed

dav1d AV1 decoder.

libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec. Requires the presence of the libdav1d headers and library during configuration. You need to explicitly configure the build with "--enable-libdav1d".

Options

The following options are supported by the libdav1d wrapper.

Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
Apply film grain to the decoded video if present in the bitstream. Defaults to the internal default of the library.
Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the internal default of the library.
Output all spatial layers of a scalable AV1 bitstream. The default value is false.

AVS2-P2/IEEE1857.4 video decoder wrapper.

This decoder allows libavcodec to decode AVS2 streams with davs2 library.

AVS3-P2/IEEE1857.10 video decoder.

libuavs3d allows libavcodec to decode AVS3 streams. Requires the presence of the libuavs3d headers and library during configuration. You need to explicitly configure the build with "--enable-libuavs3d".

Options

The following option is supported by the libuavs3d wrapper.

Set amount of frame threads to use during decoding. The default value is 0 (autodetect).

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).

AC-3 Decoder Options

Dynamic Range Scale Factor. The factor to apply to dynamic range values from the AC-3 stream. This factor is applied exponentially. The default value is 1. There are 3 notable scale factor ranges:
DRC disabled. Produces full range audio.
0 < drc_scale <= 1
DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full range and full compression.
DRC 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.

FLAC Decoder options

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".

Options

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.

ARIB STD-B24 caption decoder.

Implements profiles A and C of the ARIB STD-B24 standard.

libaribb24 Decoder Options

Sets the base path for the libaribb24 library. This is utilized for reading of configuration files (for custom unicode conversions), and for dumping of non-text symbols as images under that location.

Unset by default.

Tells the decoder wrapper to skip text blocks that contain half-height ruby text.

Enabled by default.

Options

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

Options

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 commas, 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".

Options

List of teletext page numbers to decode. Pages that do not match the specified list are dropped. You may use the special "*" string to match all pages, or "subtitle" to match all subtitle pages. Default value is *.
Set default character set used for decoding, a value between 0 and 87 (see ETS 300 706, Section 15, Table 32). Default value is -1, which does not override the libzvbi default. This option is needed for some legacy level 1.0 transmissions which cannot signal the proper charset.
Discards the top teletext line. Default value is 1.
Specifies the format of the decoded subtitles.
The default format, you should use this for teletext pages, because certain graphics and colors cannot be expressed in simple text or even ASS.
Simple text based output without formatting.
ass
Formatted ASS output, subtitle pages and teletext pages are returned in different styles, subtitle pages are stripped down to text, but an effort is made to keep the text alignment and the formatting.
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 -1 which means infinity or until the next subtitle event comes.
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.

Options

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 ac3_fixed encoder is not the default codec for any of the output formats, so it must be specified explicitly using the option "-acodec ac3_fixed" in order to use it.

AC-3 Metadata

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

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

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

Metadata Control Options

Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.
0
The metadata values set at initialization will be used for every frame in the stream. (default)
1
Metadata values can be changed before encoding each frame.

Downmix Levels

Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo. This field will only be written to the bitstream if a center channel is present. The value is specified as a scale factor. There are 3 valid values:
0.707
Apply -3dB gain
0.595
Apply -4.5dB gain (default)
0.500
Apply -6dB gain
Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo. This field will only be written to the bitstream if one or more surround channels are present. The value is specified as a scale factor. There are 3 valid values:
0.707
Apply -3dB gain
0.500
Apply -6dB gain (default)
0.000
Silence Surround Channel(s)

Audio Production Information

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

Mixing Level. 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.
0
Not Indicated (default)
1
Large Room
2
Small Room

Other Metadata Options

Copyright Indicator. Specifies whether a copyright exists for this audio.
0
No Copyright Exists (default)
1
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.
0
Not Indicated (default)
1
Not Dolby Surround Encoded
2
Dolby Surround Encoded
Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy.
0
Not Original Source
1
Original Source (default)

Extended Bitstream Information

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

Extended Bitstream Information - Part 1

Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt (Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
0
Not Indicated (default)
1
Lt/Rt Downmix Preferred
2
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.
1.414
Apply +3dB gain
1.189
Apply +1.5dB gain
1.000
Apply 0dB gain
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain (default)
0.500
Apply -6.0dB gain
0.000
Silence Center Channel
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lt/Rt mode.
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain
0.500
Apply -6.0dB gain (default)
0.000
Silence Surround Channel(s)
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lo/Ro mode.
1.414
Apply +3dB gain
1.189
Apply +1.5dB gain
1.000
Apply 0dB gain
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain (default)
0.500
Apply -6.0dB gain
0.000
Silence Center Channel
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lo/Ro mode.
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain
0.500
Apply -6.0dB gain (default)
0.000
Silence Surround Channel(s)

Extended Bitstream Information - Part 2

Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX (7.1 matrixed to 5.1). Using this option does NOT mean the encoder will actually apply Dolby Surround EX processing.
0
Not Indicated (default)
1
Dolby Surround EX Off
2
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.
0
Not Indicated (default)
1
Dolby Headphone Off
2
Dolby Headphone On
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.
0
Standard A/D Converter (default)
1
hdcd
HDCD A/D Converter

Other AC-3 Encoding Options

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.

Floating-Point-Only AC-3 Encoding Options

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

Enables/Disables use of channel coupling, which is an optional AC-3 feature that increases quality by combining high frequency information from multiple channels into a single channel. The per-channel high frequency information is sent with less accuracy in both the frequency and time domains. This allows more bits to be used for lower frequencies while preserving enough information to reconstruct the high frequencies. This option is enabled by default for the floating-point encoder and should generally be left as enabled except for testing purposes or to increase encoding speed.
-1
Selected by Encoder (default)
0
Disable Channel Coupling
1
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.
-1
Selected by Encoder (default)

FLAC (Free Lossless Audio Codec) Encoder

Options

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
2level
4level
8level
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.

Options

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 has support for the AAC-HE profiles.

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

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

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

Options

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

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.

Enable ELDv2 (LD-MPS extension for ELD stereo signals) for ELDv2 if set to 1, disabled if set to 0.

Note that option is available when fdk-aac version (AACENCODER_LIB_VL0.AACENCODER_LIB_VL1.AACENCODER_LIB_VL2) > (4.0.0).

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:

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

Default value is 0.

Examples

  • Use ffmpeg to convert an audio file to VBR AAC in an M4A (MP4) container:
    ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a
    
  • Use ffmpeg to convert an audio file to CBR 64k kbps AAC, using the High-Efficiency AAC profile:
    ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a
    

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

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

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

Options

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

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.

Options

Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
4750
5150
5900
6700
7400
7950
10200
12200
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".

Option Mapping

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

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.
Enable inband forward error correction. packet_loss must be non-zero to take advantage - frequency of FEC 'side-data' is proportional to expected packet loss. Default is disabled.
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.

Options

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".

Options

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.

Options

Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
6600
8850
12650
14250
15850
18250
19850
23050
23850
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".

Options

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

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

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.

Motion JPEG encoder.

Options

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.

Options

The equivalent options for wavpack command line utility are listed in parentheses.

Shared options

The following shared options are effective for this encoder. Only special notes about this particular encoder will be documented here. For the general meaning of the options, see the Codec Options chapter.

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.

Private options

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.

GIF

GIF image/animation encoder.

Options

Sets the flags used for GIF encoding.
Enables picture offsetting.

Default is enabled.

Enables transparency detection between frames.

Default is enabled.

Enables encoding one full GIF image per frame, rather than an animated GIF.

Default value is 0.

Writes a palette to the global GIF header where feasible.

If disabled, every frame will always have a palette written, even if there is a global palette supplied.

Default value is 1.

Vidvox Hap video encoder.

Options

format integer
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".

Options

format integer
Can be set to either "j2k" or "jp2" (the default) that makes it possible to store non-rgb pix_fmts.
Sets tile width. Range is 1 to 1073741824. Default is 256.
Sets tile height. Range is 1 to 1073741824. Default is 256.
Allows setting the discrete wavelet transform (DWT) type

Default is "dwt97int"

Enable this to add SOP marker at the start of each packet. Disabled by default.
Enable this to add EPH marker at the end of each packet header. Disabled by default.
Sets the progression order to be used by the encoder. Possible values are:

Set to "lrcp" by default.

By default, when this option is not used, compression is done using the quality metric. This option allows for compression using compression ratio. The compression ratio for each level could be specified. The compression ratio of a layer "l" species the what ratio of total file size is contained in the first "l" layers.

Example usage:

ffmpeg -i input.bmp -c:v jpeg2000 -layer_rates "100,10,1" output.j2k

This would compress the image to contain 3 layers, where the data contained in the first layer would be compressed by 1000 times, compressed by 100 in the first two layers, and shall contain all data while using all 3 layers.

rav1e AV1 encoder wrapper.

Requires the presence of the rav1e headers and library during configuration. You need to explicitly configure the build with "--enable-librav1e".

Options

Sets the maximum quantizer to use when using bitrate mode.
Sets the minimum quantizer to use when using bitrate mode.
qp
Uses quantizer mode to encode at the given quantizer (0-255).
Selects the speed preset (0-10) to encode with.
Selects how many tiles to encode with.
Selects how many rows of tiles to encode with.
Selects how many columns of tiles to encode with.
rav1e-params
Set rav1e options using a list of key=value pairs separated by ":". See rav1e --help for a list of options.

For example to specify librav1e encoding options with -rav1e-params:

ffmpeg -i input -c:v librav1e -b:v 500K -rav1e-params speed=5:low_latency=true output.mp4

libaom AV1 encoder wrapper.

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

Options

The wrapper supports the following standard libavcodec options:

Set bitrate target in bits/second. By default this will use variable-bitrate mode. If maxrate and minrate are also set to the same value then it will use constant-bitrate mode, otherwise if crf is set as well then it will use constrained-quality mode.
Set key frame placement. The GOP size sets the maximum distance between key frames; if zero the output stream will be intra-only. The minimum distance is ignored unless it is the same as the GOP size, in which case key frames will always appear at a fixed interval. Not set by default, so without this option the library has completely free choice about where to place key frames.
Set minimum/maximum quantisation values. Valid range is from 0 to 63 (warning: this does not match the quantiser values actually used by AV1 - divide by four to map real quantiser values to this range). Defaults to min/max (no constraint).
Set rate control buffering parameters. Not used if not set - defaults to unconstrained variable bitrate.
Set the number of threads to use while encoding. This may require the tiles or row-mt options to also be set to actually use the specified number of threads fully. Defaults to the number of hardware threads supported by the host machine.
Set the encoding profile. Defaults to using the profile which matches the bit depth and chroma subsampling of the input.

The wrapper also has some specific options:

Set the quality/encoding speed tradeoff. Valid range is from 0 to 8, higher numbers indicating greater speed and lower quality. The default value is 1, which will be slow and high quality.
Enable use of alternate reference frames. Defaults to the internal default of the library.
Set altref noise reduction max frame count. Default is -1.
Set altref noise reduction filter strength. Range is -1 to 6. Default is -1.
Set adaptive quantization mode. Possible values:
Disabled.
Variance-based.
Complexity-based.
Cyclic refresh.
Set the distortion metric the encoder is tuned with. Default is "psnr".
psnr (0)
ssim (1)
Set the maximum number of frames which the encoder may keep in flight at any one time for lookahead purposes. Defaults to the internal default of the library.
Enable error resilience features:
Improve resilience against losses of whole frames.

Not enabled by default.

Set the quality/size tradeoff for constant-quality (no bitrate target) and constrained-quality (with maximum bitrate target) modes. Valid range is 0 to 63, higher numbers indicating lower quality and smaller output size. Only used if set; by default only the bitrate target is used.
Set a change threshold on blocks below which they will be skipped by the encoder. Defined in arbitrary units as a nonnegative integer, defaulting to zero (no blocks are skipped).
Set a threshold for dropping frames when close to rate control bounds. Defined as a percentage of the target buffer - when the rate control buffer falls below this percentage, frames will be dropped until it has refilled above the threshold. Defaults to zero (no frames are dropped).
Amount of noise to be removed for grain synthesis. Grain synthesis is disabled if this option is not set or set to 0.
Block size used for denoising for grain synthesis. If not set, AV1 codec uses the default value of 32.
Set datarate undershoot (min) percentage of the target bitrate. Range is -1 to 100. Default is -1.
Set datarate overshoot (max) percentage of the target bitrate. Range is -1 to 1000. Default is -1.
Minimum percentage variation of the GOP bitrate from the target bitrate. If minsection-pct is not set, the libaomenc wrapper computes it as follows: "(minrate * 100 / bitrate)". Range is -1 to 100. Default is -1 (unset).
Maximum percentage variation of the GOP bitrate from the target bitrate. If maxsection-pct is not set, the libaomenc wrapper computes it as follows: "(maxrate * 100 / bitrate)". Range is -1 to 5000. Default is -1 (unset).
Enable frame parallel decodability features. Default is true.
Set the number of tiles to encode the input video with, as columns x rows. Larger numbers allow greater parallelism in both encoding and decoding, but may decrease coding efficiency. Defaults to the minimum number of tiles required by the size of the input video (this is 1x1 (that is, a single tile) for sizes up to and including 4K).
Set the number of tiles as log2 of the number of tile rows and columns. Provided for compatibility with libvpx/VP9.
Enable row based multi-threading. Disabled by default.
Enable Constrained Directional Enhancement Filter. The libaom-av1 encoder enables CDEF by default.
Enable Loop Restoration Filter. Default is true for libaom-av1.
Enable the use of global motion for block prediction. Default is true.
Enable block copy mode for intra block prediction. This mode is useful for screen content. Default is true.
Enable rectangular partitions. Default is true.
Enable 1:4/4:1 partitions. Default is true.
Enable AB shape partitions. Default is true.
Enable angle delta intra prediction. Default is true.
Enable chroma predicted from luma intra prediction. Default is true.
Enable filter intra predictor. Default is true.
Enable intra edge filter. Default is true.
Enable smooth intra prediction mode. Default is true.
Enable paeth predictor in intra prediction. Default is true.
Enable palette prediction mode. Default is true.
Enable extended transform type, including FLIPADST_DCT, DCT_FLIPADST, FLIPADST_FLIPADST, ADST_FLIPADST, FLIPADST_ADST, IDTX, V_DCT, H_DCT, V_ADST, H_ADST, V_FLIPADST, H_FLIPADST. Default is true.
Enable 64-pt transform. Default is true.
Use reduced set of transform types. Default is false.
Use DCT only for INTRA modes. Default is false.
Use DCT only for INTER modes. Default is false.
Use Default-transform only for INTRA modes. Default is false.
Enable temporal mv prediction. Default is true.
Use reduced set of single and compound references. Default is false.
Enable obmc. Default is true.
Enable dual filter. Default is true.
Enable difference-weighted compound. Default is true.
Enable distance-weighted compound. Default is true.
Enable one sided compound. Default is true.
Enable interinter wedge compound. Default is true.
Enable interintra wedge compound. Default is true.
Enable masked compound. Default is true.
Enable interintra compound. Default is true.
Enable smooth interintra mode. Default is true.
Set libaom options using a list of key=value pairs separated by ":". For a list of supported options, see aomenc --help under the section "AV1 Specific Options".

For example to specify libaom encoding options with -aom-params:

ffmpeg -i input -c:v libaom-av1 -b:v 500K -aom-params tune=psnr:enable-tpl-model=1 output.mp4

SVT-AV1 encoder wrapper.

Requires the presence of the SVT-AV1 headers and library during configuration. You need to explicitly configure the build with "--enable-libsvtav1".

Options

Set the encoding profile.
Set the operating point level.
Set the operating point tier.
Set the rate control mode to use.

Possible modes:

Constant quantizer: use fixed values of qindex (dependent on the frame type) throughout the stream. This mode is the default.
Variable bitrate: use a target bitrate for the whole stream.
Constrained variable bitrate: use a target bitrate for each GOP.
Set the maximum quantizer to use when using a bitrate mode.
Set the minimum quantizer to use when using a bitrate mode.
qp
Set the quantizer used in cqp rate control mode (0-63).
Enable scene change detection.
Set number of frames to look ahead (0-120).
Set the quality-speed tradeoff, in the range 0 to 8. Higher values are faster but lower quality. Defaults to 8 (highest speed).
Set log2 of the number of rows of tiles to use (0-6).
Set log2 of the number of columns of tiles to use (0-4).

Kvazaar H.265/HEVC encoder.

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

Options

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.

Options

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/.

Options

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.

Examples

  • Set maximum constant quality (VBR) encoding with ffmpeg:
    ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
    
  • Use ffmpeg to convert a CBR 1000 kbps Theora video stream:
    ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
    

VP8/VP9 format supported through libvpx.

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

Options

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

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

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

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)".
psnr (psnr)
ssim (ssim)
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). Values greater than 1 enable multi-layer alternate reference frames (VP9 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.
Increase sharpness at the expense of lower PSNR. The valid range is [0, 7].
Sets the temporal scalability configuration using a :-separated list of key=value pairs. For example, to specify temporal scalability parameters with "ffmpeg":
ffmpeg -i INPUT -c:v libvpx -ts-parameters ts_number_layers=3:\
ts_target_bitrate=250,500,1000:ts_rate_decimator=4,2,1:\
ts_periodicity=4:ts_layer_id=0,2,1,2:ts_layering_mode=3 OUTPUT

Below is a brief explanation of each of the parameters, please refer to "struct vpx_codec_enc_cfg" in "vpx/vpx_encoder.h" for more details.

Number of temporal coding layers.
Target bitrate for each temporal layer (in kbps). (bitrate should be inclusive of the lower temporal layer).
Frame rate decimation factor for each temporal layer.
Length of the sequence defining frame temporal layer membership.
Template defining the membership of frames to temporal layers.
(optional) Selecting the temporal structure from a set of pre-defined temporal layering modes. Currently supports the following options.
0
No temporal layering flags are provided internally, relies on flags being passed in using "metadata" field in "AVFrame" with following keys.
Sets the flags passed into the encoder to indicate the referencing scheme for the current frame. Refer to function "vpx_codec_encode" in "vpx/vpx_encoder.h" for more details.
Explicitly sets the temporal id of the current frame to encode.
2
Two temporal layers. 0-1...
3
Three temporal layers. 0-2-1-2...; with single reference frame.
4
Same as option "3", except there is a dependency between the two temporal layer 2 frames within the temporal period.
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).
colorspace color-space
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.

Enable temporal dependency model.
Using per-frame metadata, set members of the structure "vpx_svc_ref_frame_config_t" in "vpx/vp8cx.h" to fine-control referencing schemes and frame buffer management. Use a :-separated list of key=value pairs. For example,
av_dict_set(&av_frame->metadata, "ref-frame-config", \
"rfc_update_buffer_slot=7:rfc_lst_fb_idx=0:rfc_gld_fb_idx=1:rfc_alt_fb_idx=2:rfc_reference_last=0:rfc_reference_golden=0:rfc_reference_alt_ref=0");
Indicates the buffer slot number to update
Indicates whether to update the LAST frame
Indicates whether to update GOLDEN frame
Indicates whether to update ALT_REF frame
LAST frame buffer index
GOLDEN frame buffer index
ALT_REF frame buffer index
Indicates whether to reference LAST frame
Indicates whether to reference GOLDEN frame
Indicates whether to reference ALT_REF frame
Indicates frame duration

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.

Pixel Format

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

Options

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.

Supported Pixel Formats

x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at x264's configure time.

Options

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

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

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

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.
qp (qp)
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
ssim (ssim)
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.
8x8dct
Enable adaptive spatial transform (high profile 8x8 transform) when set to 1. When set to 0, it has the same effect as x264's --no-8x8dct option.
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.
deblock (deblock)
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).
x264-params (N.A.)
Override the x264 configuration using a :-separated list of key=value parameters.

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

For example to specify libx264 encoding options with ffmpeg:

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

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

x265 H.265/HEVC encoder wrapper.

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

Options

Sets target video bitrate.
Set the GOP size.
Minimum GOP size.
Number of reference frames each P-frame can use. The range is from 1-16.
Set the x265 preset.
Set the x265 tune parameter.
Set profile restrictions.
Set the quality for constant quality mode.
qp
Set constant quantization rate control method parameter.
Minimum quantizer scale.
Maximum quantizer scale.
Maximum difference between quantizer scales.
Quantizer curve blur
Quantizer curve compression factor
Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces it to choose an IDR-frame.
x265-params
Set x265 options using a list of key=value couples separated by ":". See x265 --help for a list of options.

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

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

xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.

This encoder requires the presence of the libxavs2 headers and library during configuration. You need to explicitly configure the build with --enable-libxavs2.

The following standard libavcodec options are used:

  • b / bit_rate
  • g / gop_size
  • bf / max_b_frames

The encoder also has its own specific options:

Options

Set the number of parallel threads for rows from 1 to 8 (default 5).
Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set the initial qp for the first frame.
qp
Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set the qp value under constant-QP mode.
Set the max qp for rate control from 1 to 63 (default 55).
Set the min qp for rate control from 1 to 63 (default 20).
Set the Speed level from 0 to 9 (default 0). Higher is better but slower.
Set the log level from -1 to 3 (default 0). -1: none, 0: error, 1: warning, 2: info, 3: debug.
xavs2-params
Set xavs2 options using a list of key=value couples separated by ":".

For example to specify libxavs2 encoding options with -xavs2-params:

ffmpeg -i input -c:v libxavs2 -xavs2-params RdoqLevel=0 output.avs2

Xvid MPEG-4 Part 2 encoder wrapper.

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

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

Options

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

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.

ssim
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.

This provides wrappers to encoders (both audio and video) in the MediaFoundation framework. It can access both SW and HW encoders. Video encoders can take input in either of nv12 or yuv420p form (some encoders support both, some support only either - in practice, nv12 is the safer choice, especially among HW encoders).

MPEG-2 video encoder.

Options

Select the mpeg2 profile to encode:
422
Spatially Scalable
SNR Scalable
Select the mpeg2 level to encode:
Specifies if the encoder should write a sequence_display_extension to the output.
-1
Decide automatically to write it or not (this is the default) by checking if the data to be written is different from the default or unspecified values.
0
Never write it.
1
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.
Import closed captions (which must be ATSC compatible format) into output. Default is 1 (on).

PNG image encoder.

Private options

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.

Private Options for prores-ks

Select the ProRes profile to encode
4444
4444xq
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.

Speed considerations

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

Setting a higher bits_per_mb limit will improve the speed.

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

QSV encoders

The family of Intel QuickSync Video encoders (MPEG-2, H.264, HEVC, JPEG/MJPEG and VP9)

The ratecontrol method is selected as follows:

When global_quality is specified, a quality-based mode is used. Specifically this means either
  • CQP - constant quantizer scale, when the qscale codec flag is also set (the -qscale ffmpeg option).
  • LA_ICQ - intelligent constant quality with lookahead, when the look_ahead option is also set.
  • ICQ -- intelligent constant quality otherwise.
Otherwise, a bitrate-based mode is used. For all of those, you should specify at least the desired average bitrate with the b option.
  • LA - VBR with lookahead, when the look_ahead option is specified.
  • VCM - video conferencing mode, when the vcm option is set.
  • CBR - constant bitrate, when maxrate is specified and equal to the average bitrate.
  • VBR - variable bitrate, when maxrate is specified, but is higher than the average bitrate.
  • AVBR - average VBR mode, when maxrate is not specified. This mode is further configured by the avbr_accuracy and avbr_convergence options.

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

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

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

Options

dia size for the iterative motion estimation

VAAPI encoders

Wrappers for hardware encoders accessible via VAAPI.

These encoders only accept input in VAAPI hardware surfaces. If you have input in software frames, use the hwupload filter to upload them to the GPU.

The following standard libavcodec options are used:

  • g / gop_size
  • bf / max_b_frames
  • profile

    If not set, this will be determined automatically from the format of the input frames and the profiles supported by the driver.

  • level
  • b / bit_rate
  • maxrate / rc_max_rate
  • bufsize / rc_buffer_size
  • rc_init_occupancy / rc_initial_buffer_occupancy
  • compression_level

    Speed / quality tradeoff: higher values are faster / worse quality.

  • q / global_quality

    Size / quality tradeoff: higher values are smaller / worse quality.

  • qmin
  • qmax
  • i_qfactor / i_quant_factor
  • i_qoffset / i_quant_offset
  • b_qfactor / b_quant_factor
  • b_qoffset / b_quant_offset
  • slices

All encoders support the following options:

Some drivers/platforms offer a second encoder for some codecs intended to use less power than the default encoder; setting this option will attempt to use that encoder. Note that it may support a reduced feature set, so some other options may not be available in this mode.
Set the number of normal intra frames between full-refresh (IDR) frames in open-GOP mode. The intra frames are still IRAPs, but will not include global headers and may have non-decodable leading pictures.
Set the B-frame reference depth. When set to one (the default), all B-frames will refer only to P- or I-frames. When set to greater values multiple layers of B-frames will be present, frames in each layer only referring to frames in higher layers.
Set the rate control mode to use. A given driver may only support a subset of modes.

Possible modes:

Choose the mode automatically based on driver support and the other options. This is the default.
Constant-quality.
Constant-bitrate.
Variable-bitrate.
Intelligent constant-quality.
Quality-defined variable-bitrate.
Average variable bitrate.

Each encoder also has its own specific options:

profile sets the value of profile_idc and the constraint_set*_flags. level sets the value of level_idc.
Set entropy encoder (default is cabac). Possible values:
Use CABAC.
Use CAVLC.
Include access unit delimiters in the stream (not included by default).
Set SEI message types to include. Some combination of the following values:
Include a user_data_unregistered message containing information about the encoder.
Include picture timing parameters (buffering_period and pic_timing messages).
Include recovery points where appropriate (recovery_point messages).
profile and level set the values of general_profile_idc and general_level_idc respectively.
Include access unit delimiters in the stream (not included by default).
Set general_tier_flag. This may affect the level chosen for the stream if it is not explicitly specified.
Set SEI message types to include. Some combination of the following values:
Include HDR metadata if the input frames have it (mastering_display_colour_volume and content_light_level messages).
Set the number of tiles to encode the input video with, as columns x rows. Larger numbers allow greater parallelism in both encoding and decoding, but may decrease coding efficiency.
Only baseline DCT encoding is supported. The encoder always uses the standard quantisation and huffman tables - global_quality scales the standard quantisation table (range 1-100).

For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported. RGB is also supported, and will create an RGB JPEG.

Include JFIF header in each frame (not included by default).
Include standard huffman tables (on by default). Turning this off will save a few hundred bytes in each output frame, but may lose compatibility with some JPEG decoders which don't fully handle MJPEG.
profile and level set the value of profile_and_level_indication.
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).

Options

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

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

Options

Specify the global palette used by the bitmaps.

The format for this option is a string containing 16 24-bits hexadecimal numbers (without 0x prefix) separated by commas, for example "0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b".

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.

Modify metadata embedded in an AV1 stream.

Insert or remove temporal delimiter OBUs in all temporal units of the stream.
Insert a TD at the beginning of every TU which does not already have one.
Remove the TD from the beginning of every TU which has one.
Set the color description fields in the stream (see AV1 section 6.4.2).
Set the color range in the stream (see AV1 section 6.4.2; note that this cannot be set for streams using BT.709 primaries, sRGB transfer characteristic and identity (RGB) matrix coefficients).
Limited range.
Full range.
Set the chroma sample location in the stream (see AV1 section 6.4.2). This can only be set for 4:2:0 streams.
Left position (matching the default in MPEG-2 and H.264).
Top-left position.
Set the tick rate (num_units_in_display_tick / time_scale) in the timing info in the sequence header.
Set the number of ticks in each picture, to indicate that the stream has a fixed framerate. Ignored if tick_rate is not also set.
Deletes Padding OBUs.

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 except when said packets already exactly begin with the extradata that is intended to be added.

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 k.

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

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

Extract 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 whether the stream is suitable for display using overscan or not (see H.264 section E.2.1).
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.
Set the level in the SPS. Refer to H.264 section A.3 and tables A-1 to A-5.

The argument must be the name of a level (for example, 4.2), a level_idc value (for example, 42), or the special name auto indicating that the filter should attempt to guess the level from the input stream properties.

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).

Set the level in the VPS and SPS. See H.265 section A.4 and tables A.6 and A.7.

The argument must be the name of a level (for example, 5.1), a general_level_idc value (for example, 153 for level 5.1), or the special name auto indicating that the filter should attempt to guess the level from the input stream properties.

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:

4/3
16/9
221/100

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.

Repacketize PCM audio to a fixed number of samples per packet or a fixed packet rate per second. This is similar to the asetnsamples audio filter but works on audio packets instead of audio frames.

Set the number of samples per each output audio packet. 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 packet with silence, so that it will contain the same number of samples (or roughly the same number of samples, see frame_rate) as the previous ones. Default value is 1.
This option makes the filter output a fixed number of packets per second instead of a fixed number of samples per packet. If the audio sample rate is not divisible by the frame rate then the number of samples will not be constant but will vary slightly so that each packet will start as close to the frame boundary as possible. Using this option has precedence over nb_out_samples.

You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio for NTSC frame rate using the frame_rate option.

ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -

Modify color property metadata embedded in prores stream.

Set the color primaries. Available values are:
Keep the same color primaries property (default).
BT601 625
BT601 525
DCI P3
P3 D65
Set the color transfer. Available values are:
Keep the same transfer characteristics property (default).
BT 601, BT 709, BT 2020
SMPTE ST 2084
ARIB STD-B67
Set the matrix coefficient. Available values are:
Keep the same colorspace property (default).
BT 601

Set Rec709 colorspace for each frame of the file

ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov

Set Hybrid Log-Gamma parameters for each frame of the file

ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov

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.

Set PTS and DTS in packets.

It accepts the following parameters:

Set expressions for PTS, DTS or both.

The expressions are evaluated through the eval API and can contain the following constants:

The count of the input packet. Starting from 0.
The demux timestamp in input in case of "ts" or "dts" option or presentation timestamp in case of "pts" option.
The original position in the file of the packet, or undefined if undefined for the current packet
The demux timestamp in input.
The presentation timestamp in input.
The DTS of the first packet.
The PTS of the first packet.
The previous input DTS.
The previous input PTS.
The previous output DTS.
The previous output PTS.
The timebase of stream packet belongs.
The sample rate of stream packet belongs.

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 AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending on the build only a subset of these may be available.

Extract the core from a TrueHD stream, dropping ATMOS data.

Modify metadata embedded in a VP9 stream.

Set the color space value in the frame header. Note that any frame set to RGB will be implicitly set to PC range and that RGB is incompatible with profiles 0 and 2.
Set the color range value in the frame header. Note that any value imposed by the color space will take precedence over this value.

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 the maximum number of buffered packets when probing a codec. Default is 2500 packets.
Set packet size.
Set format flags. Some are implemented for a limited number of formats.

Possible values for input files:

Discard corrupted packets.
Enable fast, but inaccurate seeks for some formats.
Generate missing PTS if DTS is present.
Ignore DTS if PTS is set. Inert when nofillin is set.
Ignore index.
Reduce the latency introduced by buffering during initial input streams analysis.
Do not fill in missing values in packet fields that can be exactly calculated.
Disable AVParsers, this needs "+nofillin" too.
Try to interleave output packets by DTS. At present, available only for AVIs with an index.

Possible values for output files:

Automatically apply bitstream filters as required by the output format. Enabled by default.
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.
Write out packets immediately.
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.
buffer
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 10000000 (10 seconds).

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 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 indentation:
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.
Skip estimation of input duration when calculated using PTS. At present, applicable for MPEG-PS and MPEG-TS.
Specify how strictly to follow the standards. "f_strict" is deprecated and should be used only via the ffmpeg tool.

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.

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

The exact semantics of stream specifiers is defined by the "avformat_match_stream_specifier()" function declared in the libavformat/avformat.h header and documented in the Stream specifiers section in the ffmpeg(1) manual.

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.

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.

Syntax

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

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

All subsequent file-related directives apply to that file.

"ffconcat version 1.0"
Identify the script type and version. It also sets the safe option to 1 if it was -1.

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

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

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

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

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

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

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

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

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

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

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

"file_packet_metadata key=value"
Metadata of the packets of the file. The specified metadata will be set for each file packet. You can specify this directive multiple times to add multiple metadata entries.
"stream"
Introduce a stream in the virtual file. All subsequent stream-related directives apply to the last introduced stream. Some streams properties must be set in order to allow identifying the matching streams in the subfiles. If no streams are defined in the script, the streams from the first file are copied.
"exact_stream_id id"
Set the id of the stream. If this directive is given, the string with the corresponding id in the subfiles will be used. This is especially useful for MPEG-PS (VOB) files, where the order of the streams is not reliable.

Options

This demuxer accepts the following option:

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.

Examples

  • Use absolute filenames and include some comments:
    # my first filename
    file /mnt/share/file-1.wav
    # my second filename including whitespace
    file '/mnt/share/file 2.wav'
    # my third filename including whitespace plus single quote
    file '/mnt/share/file 3'\''.wav'
    
  • Allow for input format auto-probing, use safe filenames and set the duration of the first file:
    ffconcat version 1.0
    
    file file-1.wav
    duration 20.0
    
    file subdir/file-2.wav
    

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.
Ignore the size of previous tag value.
Output all context of the onMetadata.

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

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".

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.
The maximum number of times to load m3u8 when it refreshes without new segments. 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.
Use HTTP partial requests for downloading HTTP segments. 0 = disable, 1 = enable, -1 = auto, Default is auto.

Image file demuxer.

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

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

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

This demuxer accepts the following options:

framerate
Set the frame rate for the video stream. It defaults to 25.
loop
If set to 1, loop over the input. Default value is 0.
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.
If set to 1, will add two extra fields to the metadata found in input, making them also available for other filters (see drawtext filter for examples). Default value is 0. The extra fields are described below:
Corresponds to the full path to the input file being read.
Corresponds to the name of the file being read.

Examples

  • Use ffmpeg for creating a video from the images in the file sequence img-001.jpeg, img-002.jpeg, ..., assuming an input frame rate of 10 frames per second:
    ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
    
  • As above, but start by reading from a file with index 100 in the sequence:
    ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
    
  • Read images matching the "*.png" glob pattern , that is all the files terminating with the ".png" suffix:
    ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
    

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

See https://bitbucket.org/mpyne/game-music-emu/overview for more information.

It accepts the following options:

Set the index of which track to demux. The demuxer can only export one track. Track indexes start at 0. Default is to pick the first track. Number of tracks is exported as tracks metadata entry.
Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of files that can be read. Default is 50 MiB.

ModPlug based module demuxer

See https://github.com/Konstanty/libmodplug

It will export one 2-channel 16-bit 44.1 kHz audio stream. Optionally, a "pal8" 16-color video stream can be exported with or without printed metadata.

It accepts the following options:

Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.
Set amount of reverb. Range 0-100. Default is 0.
Set delay in ms, clamped to 40-250 ms. Default is 0.
Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.
Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.
Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.
Set surround delay in ms, clamped to 5-40 ms. Default is 0.
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB. 0 removes buffer size limit (not recommended). Default is 5 MiB.
String which is evaluated using the eval API to assign colors to the generated video stream. Variables which can be used are "x", "y", "w", "h", "t", "speed", "tempo", "order", "pattern" and "row".
Generate video stream. Can be 1 (on) or 0 (off). Default is 0.
Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
Print metadata on video stream. Includes "speed", "tempo", "order", "pattern", "row" and "ts" (time in ms). Can be 1 (on) or 0 (off). Default is 1.

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.

Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).

Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v

Options

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.
When seeking, identify the closest point in each stream individually and demux packets in that stream from identified point. This can lead to a different sequence of packets compared to demuxing linearly from the beginning. Default is true.
Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the timeline described by the edit list. Default is false.
Modify the stream index to reflect the timeline described by the edit list. "ignore_editlist" must be set to false for this option to be effective. If both "ignore_editlist" and this option are set to false, then only the start of the stream index is modified to reflect initial dwell time or starting timestamp described by the edit list. Default is true.
Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are only parsed when input is seekable. Default is false.
For seekable fragmented input, set fragment's starting timestamp from media fragment random access box, if present.

Following options are available:

Auto-detect whether to set mfra timestamps as PTS or DTS (default)
Set mfra timestamps as DTS
Set mfra timestamps as PTS
0
Don't use mfra box to set timestamps
Export unrecognized boxes within the udta box as metadata entries. The first four characters of the box type are set as the key. Default is false.
Export entire contents of XMP_ box and uuid box as a string with key "xmp". Note that if "export_all" is set and this option isn't, the contents of XMP_ box are still exported but with key "XMP_". Default is false.
4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.
Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to specify.
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).

Audible AAX

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

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

MPEG-2 transport stream demuxer.

This demuxer accepts the following options:

Set size limit for looking up a new synchronization. Default value is 65536.
Skip PMTs for programs not defined in the PAT. Default value is 0.
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.
Re-use existing streams when a PMT's version is updated and elementary streams move to different PIDs. Default value is 0.

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:

framerate
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 http://www.ted.com/.

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

This demuxer accepts the following option:

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

Vapoursynth wrapper.

Due to security concerns, Vapoursynth scripts will not be autodetected so the input format has to be forced. For ff* CLI tools, add "-f vapoursynth" before the input "-i yourscript.vpy".

This demuxer accepts the following option:

The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of scripts that can be read. Default is 1 MiB.

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.

Options

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.

Options

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.

Options

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 the "amerge" section in the ffmpeg-filters manual).

If set to true, store positive height for raw RGB bitmaps, which indicates bitmap is stored bottom-up. Note that this option does not flip the bitmap which has to be done manually beforehand, e.g. by using the vflip filter. Default is false and indicates bitmap is stored top down.

Chromaprint fingerprinter.

This muxer feeds audio data to the Chromaprint library, which generates a fingerprint for the provided audio data. See https://acoustid.org/chromaprint

It takes a single signed native-endian 16-bit raw audio stream of at most 2 channels.

Options

Threshold for detecting silence. Range is from -1 to 32767, where -1 disables silence detection. Silence detection can only be used with version 3 of the algorithm. Silence detection must be disabled for use with the AcoustID service. Default is -1.
Version of algorithm to fingerprint with. Range is 0 to 4. Version 3 enables silence detection. Default is 1.
Format to output the fingerprint as. Accepts the following options:
Binary raw fingerprint
Binary compressed fingerprint
Base64 compressed fingerprint (default)

CRC (Cyclic Redundancy Check) testing format.

This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.

The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames.

See also the framecrc muxer.

Examples

For example to compute the CRC of the input, and store it in the file out.crc:

ffmpeg -i INPUT -f crc out.crc

You can print the CRC to stdout with the command:

ffmpeg -i INPUT -f crc -

You can select the output format of each frame with ffmpeg by specifying the audio and video codec and format. For example to compute the CRC of the input audio converted to PCM unsigned 8-bit and the input video converted to MPEG-2 video, use the command:

ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -

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 "$Time$". In addition to the standard identifiers, an ffmpeg-specific "$ext$" identifier is also supported. When specified ffmpeg will replace $ext$ in the file name with muxing format's extensions such as mp4, webm etc.,

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
This is a deprecated option to set the segment length in microseconds, use seg_duration instead.
Set the segment length in seconds (fractional value can be set). The value is treated as average segment duration when use_template is enabled and use_timeline is disabled and as minimum segment duration for all the other use cases.
Set the length in seconds of fragments within segments (fractional value can be set).
Set the type of interval for fragmentation.
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". In the template, "$ext$" is replaced with the file name extension specific for the segment format.
DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format.
DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format.
URL of the page that will return the UTC timestamp in ISO format. Example: "https://time.akamai.com/?iso"
Use the given HTTP method to create output files. Generally set to PUT or POST.
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 hls_master_name. One media playlist file is generated for each stream with filenames media_0.m3u8, media_1.m3u8, etc.
HLS master playlist name. Default is "master.m3u8".
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.

Optional syntax is "id=x,seg_duration=x,frag_duration=x,frag_type=type,descriptor=descriptor_string,streams=a,b,c id=y,seg_duration=y,frag_type=type,streams=d,e" and so on, descriptor is useful to the scheme defined by ISO/IEC 23009-1:2014/Amd.2:2015. For example, -adaptation_sets "id=0,descriptor=<SupplementalProperty schemeIdUri=\"urn:mpeg:dash:srd:2014\" value=\"0,0,0,1,1,2,2\"/>,streams=v". Please note that descriptor string should be a self-closing xml tag. seg_duration, frag_duration and frag_type override the global option values for each adaptation set. For example, -adaptation_sets "id=0,seg_duration=2,frag_duration=1,frag_type=duration,streams=v id=1,seg_duration=2,frag_type=none,streams=a" type_id marks an adaptation set as containing streams meant to be used for Trick Mode for the referenced adaptation set. For example, -adaptation_sets "id=0,seg_duration=2,frag_type=none,streams=0 id=1,seg_duration=10,frag_type=none,trick_id=0,streams=1"

Set timeout for socket I/O operations. Applicable only for HTTP output.
Enable (1) or Disable (0) segment index correction logic. Applicable only when use_template is enabled and use_timeline is disabled.

When enabled, the logic monitors the flow of segment indexes. If a streams's segment index value is not at the expected real time position, then the logic corrects that index value.

Typically this logic is needed in live streaming use cases. The network bandwidth fluctuations are common during long run streaming. Each fluctuation can cause the segment indexes fall behind the expected real time position.

Set container format (mp4/webm) options using a ":" separated list of key=value parameters. Values containing ":" special characters must be escaped.
Write global SIDX atom. Applicable only for single file, mp4 output, non-streaming mode.
Possible values:
If this flag is set, the dash segment files format will be selected based on the stream codec. This is the default mode.
If this flag is set, the dash segment files will be in in ISOBMFF format.
If this flag is set, the dash segment files will be in in WebM format.
Ignore IO errors during open and write. Useful for long-duration runs with network output.
Enable Low-latency HLS(LHLS). Adds #EXT-X-PREFETCH tag with current segment's URI. Apple doesn't have an official spec for LHLS. Meanwhile hls.js player folks are trying to standardize a open LHLS spec. The draft spec is available in https://github.com/video-dev/hlsjs-rfcs/blob/lhls-spec/proposals/0001-lhls.md This option will also try to comply with the above open spec, till Apple's spec officially supports it. Applicable only when streaming and hls_playlist options are enabled. This is an experimental feature.
Enable Low-latency Dash by constraining the presence and values of some elements.
Publish master playlist repeatedly every after specified number of segment intervals.
Write Producer Reference Time elements on supported streams. This also enables writing prft boxes in the underlying muxer. Applicable only when the utc_url option is enabled. It's set to auto by default, in which case the muxer will attempt to enable it only in modes that require it.
Set one or more manifest profiles.
A :-separated list of key=value options to pass to the underlying HTTP protocol. Applicable only for HTTP output.
Set an intended target latency in seconds (fractional value can be set) for serving. Applicable only when streaming and write_prft options are enabled. This is an informative fields clients can use to measure the latency of the service.
Set the minimum playback rate indicated as appropriate for the purposes of automatically adjusting playback latency and buffer occupancy during normal playback by clients.
Set the maximum playback rate indicated as appropriate for the purposes of automatically adjusting playback latency and buffer occupancy during normal playback by clients.
Set the mpd update period ,for dynamic content.
The unit is second.

Per-packet CRC (Cyclic Redundancy Check) testing format.

This muxer computes and prints the Adler-32 CRC for each audio and video packet. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.

The output of the muxer consists of a line for each audio and video packet of the form:

<stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, 0x<CRC>

CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the packet.

Examples

For example to compute the CRC of the audio and video frames in INPUT, converted to raw audio and video packets, and store it in the file out.crc:

ffmpeg -i INPUT -f framecrc out.crc

To print the information to stdout, use the command:

ffmpeg -i INPUT -f framecrc -

With ffmpeg, you can select the output format to which the audio and video frames are encoded before computing the CRC for each packet by specifying the audio and video codec. For example, to compute the CRC of each decoded input audio frame converted to PCM unsigned 8-bit and of each decoded input video frame converted to MPEG-2 video, use the command:

ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -

See also the crc muxer.

Per-packet hash testing format.

This muxer computes and prints a cryptographic hash for each audio and video packet. This can be used for packet-by-packet equality checks without having to individually do a binary comparison on each.

By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash, but the output of explicit conversions to other codecs can also be used. It uses the SHA-256 cryptographic hash function by default, but supports several other algorithms.

The output of the muxer consists of a line for each audio and video packet of the form:

<stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, <hash>

hash is a hexadecimal number representing the computed hash for the packet.

hash algorithm
Use the cryptographic hash function specified by the string algorithm. Supported values include "MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256" (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32" and "adler32".

Examples

To compute the SHA-256 hash of the audio and video frames in INPUT, converted to raw audio and video packets, and store it in the file out.sha256:

ffmpeg -i INPUT -f framehash out.sha256

To print the information to stdout, using the MD5 hash function, use the command:

ffmpeg -i INPUT -f framehash -hash md5 -

See also the hash muxer.

Per-packet MD5 testing format.

This is a variant of the framehash muxer. Unlike that muxer, it defaults to using the MD5 hash function.

Examples

To compute the MD5 hash of the audio and video frames in INPUT, converted to raw audio and video packets, and store it in the file out.md5:

ffmpeg -i INPUT -f framemd5 out.md5

To print the information to stdout, use the command:

ffmpeg -i INPUT -f framemd5 -

See also the framehash and md5 muxers.

Animated GIF muxer.

It accepts the following options:

loop
Set the number of times to loop the output. Use "-1" for no loop, 0 for looping indefinitely (default).
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.

hash algorithm
Use the cryptographic hash function specified by the string algorithm. Supported values include "MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256" (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32" and "adler32".

Examples

To compute the SHA-256 hash of the input converted to raw audio and video, and store it in the file out.sha256:

ffmpeg -i INPUT -f hash out.sha256

To print an MD5 hash to stdout use the command:

ffmpeg -i INPUT -f hash -hash md5 -

See also the framehash muxer.

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.

Options

This muxer supports the following options:

Set the initial target segment length. Default value is 0.

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

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. Default value is 2.

duration must be a time duration specification, see the Time duration section in the ffmpeg-utils(1) manual. 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 the microseconds 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 strftime_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, but only in one of them. (Additionally, %v may appear multiple times in the last sub-directory or filename.) If the string %v 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.

Same as strftime option, will be deprecated.
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 -strftime 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 -strftime 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.

Same as strftime_mkdir option, will be deprecated .
Used together with -strftime_mkdir, it will create all subdirectories which is expanded in filename.
ffmpeg -i in.nut -strftime 1 -strftime_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 -strftime 1 -strftime_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.
16-octet 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.
16-octet initialization vector for every segment instead of the autogenerated ones.
Possible values:
mpegts
Output segment files in MPEG-2 Transport Stream format. This is compatible with all HLS versions.
Output segment files in fragmented MP4 format, similar to MPEG-DASH. fmp4 files may be used in HLS version 7 and above.
Set filename to the fragment files header file, default filename is init.mp4.

Use "-strftime 1" on filename to expand the segment filename with localtime.

ffmpeg -i in.nut  -hls_segment_type fmp4 -strftime 1 -hls_fmp4_init_filename "%s_init.mp4" out.m3u8

This will produce init like this 1602678741_init.mp4

Resend init file after m3u8 file refresh every time, default is 0.

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.
Add the "#EXT-X-I-FRAMES-ONLY" to playlists that has video segments and can play only I-frames in the "#EXT-X-BYTERANGE" mode.
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 strftime 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 strftime 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 strftime 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 \
   -strftime 1 -strftime_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. This flag also affects how m3u8 playlist files are created. If this flag is set, all playlist files will written into temporary file and renamed after they are complete, similarly as segments are handled. But playlists with "file" protocol and with type ("hls_playlist_type") other than "vod" are always written into temporary file regardless of this flag. Master playlist files ("master_pl_name"), if any, with "file" protocol, are always written into temporary file regardless of this flag if "master_pl_publish_rate" value is other than zero.
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. If you want something meaningful text instead of indexes in result names, you may specify names for each or some of the variants as in the following example.

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,name:my_hd v:1,a:1,name:my_sd" \
  http://example.com/live/out_%v.m3u8

This example creates two hls variant streams as in the previous one. But here, the two media playlist with file names out_my_hd.m3u8 and out_my_sd.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.

ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
  -map 0:a -map 0:a -map 0:v -f hls \
  -var_stream_map "a:0,agroup:aud_low,default:yes a:1,agroup:aud_low v:0,agroup:aud_low" \
  -master_pl_name master.m3u8 \
  http://example.com/live/out_%v.m3u8

This example creates two audio only and one 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 one video only variant streams with audio group name 'aud_low', and the audio group have default stat is NO or YES.

By default, a single hls variant containing all the encoded streams is created.

ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
  -map 0:a -map 0:a -map 0:v -f hls \
  -var_stream_map "a:0,agroup:aud_low,default:yes,language:ENG a:1,agroup:aud_low,language:CHN v:0,agroup:aud_low" \
  -master_pl_name master.m3u8 \
  http://example.com/live/out_%v.m3u8

This example creates two audio only and one 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 one video only variant streams with audio group name 'aud_low', and the audio group have default stat is NO or YES, and one audio have and language is named ENG, the other audio language is named CHN.

By default, a single hls variant containing all the encoded streams is created.

ffmpeg -y -i input_with_subtitle.mkv \
 -b:v:0 5250k -c:v h264 -pix_fmt yuv420p -profile:v main -level 4.1 \
 -b:a:0 256k \
 -c:s webvtt -c:a mp2 -ar 48000 -ac 2 -map 0:v -map 0:a:0 -map 0:s:0 \
 -f hls -var_stream_map "v:0,a:0,s:0,sgroup:subtitle" \
 -master_pl_name master.m3u8 -t 300 -hls_time 10 -hls_init_time 4 -hls_list_size \
 10 -master_pl_publish_rate 10  -hls_flags \
 delete_segments+discont_start+split_by_time ./tmp/video.m3u8

This example adds "#EXT-X-MEDIA" tag with "TYPE=SUBTITLES" in the master playlist with webvtt subtitle group name 'subtitle'. Please make sure the input file has one text subtitle stream at least.

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', language '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.
Ignore IO errors during open, write and delete. Useful for long-duration runs with network output.
Set custom HTTP headers, can override built in default headers. Applicable only for HTTP output.

ICO file muxer.

Microsoft's icon file format (ICO) has some strict limitations that should be noted:

  • Size cannot exceed 256 pixels in any dimension
  • Only BMP and PNG images can be stored
  • If a BMP image is used, it must be one of the following pixel formats:
    BMP Bit Depth      FFmpeg Pixel Format
    1bit               pal8
    4bit               pal8
    8bit               pal8
    16bit              rgb555le
    24bit              bgr24
    32bit              bgra
    
  • If a BMP image is used, it must use the BITMAPINFOHEADER DIB header
  • If a PNG image is used, it must use the rgba pixel format

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 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.

Options

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.
Set protocol options as a :-separated list of key=value parameters. Values containing the ":" special character must be escaped.

Examples

The following example shows how to use ffmpeg for creating a sequence of files img-001.jpeg, img-002.jpeg, ..., taking one image every second from the input video:

ffmpeg -i in.avi -vsync cfr -r 1 -f image2 'img-%03d.jpeg'

Note that with ffmpeg, if the format is not specified with the "-f" option and the output filename specifies an image file format, the image2 muxer is automatically selected, so the previous command can be written as:

ffmpeg -i in.avi -vsync cfr -r 1 'img-%03d.jpeg'

Note also that the pattern must not necessarily contain "%d" or "%0Nd", for example to create a single image file img.jpeg from the start of the input video you can employ the command:

ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg

The strftime option allows you to expand the filename with date and time information. Check the documentation of the "strftime()" function for the syntax.

For example to generate image files from the "strftime()" "%Y-%m-%d_%H-%M-%S" pattern, the following ffmpeg command can be used:

ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"

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"

A more complex example is to publish contents of your desktop directly to a WebDAV server every second:

ffmpeg -f x11grab -framerate 1 -i :0.0 -q:v 6 -update 1 -protocol_opts method=PUT http://example.com/desktop.jpg

Matroska container muxer.

This muxer implements the matroska and webm container specs.

Metadata

The recognized metadata settings in this muxer are:

Set title name provided to a single track. This gets mapped to the FileDescription element for a stream written as attachment.
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

Options

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 reserved space does not suffice, no Cues will be written, the file will be finalized and writing the trailer will return an error. 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.

This option controls how the FlagDefault of the output tracks will be set. It influences which tracks players should play by default. The default mode is infer.
In this mode, for each type of track (audio, video or subtitle), if there is a track with disposition default of this type, then the first such track (i.e. the one with the lowest index) will be marked as default; if no such track exists, the first track of this type will be marked as default instead (if existing). This ensures that the default flag is set in a sensible way even if the input originated from containers that lack the concept of default tracks.
This mode is the same as infer except that if no subtitle track with disposition default exists, no subtitle track will be marked as default.
In this mode the FlagDefault is set if and only if the AV_DISPOSITION_DEFAULT flag is set in the disposition of the corresponding stream.
If set to true, store positive height for raw RGB bitmaps, which indicates bitmap is stored bottom-up. Note that this option does not flip the bitmap which has to be done manually beforehand, e.g. by using the vflip filter. Default is false and indicates bitmap is stored top down.

MD5 testing format.

This is a variant of the hash muxer. Unlike that muxer, it defaults to using the MD5 hash function.

Examples

To compute the MD5 hash of the input converted to raw audio and video, and store it in the file out.md5:

ffmpeg -i INPUT -f md5 out.md5

You can print the MD5 to stdout with the command:

ffmpeg -i INPUT -f md5 -

See also the hash and framemd5 muxers.

MOV/MP4/ISMV (Smooth Streaming) muxer.

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

Options

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

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.

Skip writing of sidx atom. When bitrate overhead due to sidx atom is high, this option could be used for cases where sidx atom is not mandatory. When global_sidx flag is enabled, this option will be ignored.
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.

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

Write producer time reference box (PRFT) with a specified time source for the NTP field in the PRFT box. Set value as wallclock to specify timesource as wallclock time and pts to specify timesource as input packets' PTS values.

Setting value to pts is applicable only for a live encoding use case, where PTS values are set as as wallclock time at the source. For example, an encoding use case with decklink capture source where video_pts and audio_pts are set to abs_wallclock.

Example

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

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

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

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

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

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

  • A Xing/LAME frame right after the ID3v2 header (if present). It is enabled by default, but will be written only if the output is seekable. The "write_xing" private option can be used to disable it. The frame contains various information that may be useful to the decoder, like the audio duration or encoder delay.
  • A legacy ID3v1 tag at the end of the file (disabled by default). It may be enabled with the "write_id3v1" private option, but as its capabilities are very limited, its usage is not recommended.

Examples:

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

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

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

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

Write a "clean" MP3 without any extra features:

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

MPEG transport stream muxer.

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

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

Options

The muxer options are:

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 hexadecimal value between 0x01 and 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 PMTs. Default is 0x1000, minimum is 0x0020, maximum is 0x1ffa. This option has no effect in m2ts mode where the PMT PID is fixed 0x0100.
Set the first PID for elementary streams. Default is 0x0100, minimum is 0x0020, maximum is 0x1ffa. This option has no effect in m2ts mode where the elementary stream PIDs are fixed.
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.
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. Default is "-1" which means that the PCR interval will be determined automatically: 20 ms is used for CBR streams, the highest multiple of the frame duration which is less than 100 ms is used for VBR streams.
Maximum time in seconds between PAT/PMT tables. Default is 0.1.
Maximum time in seconds between SDT tables. Default is 0.5.
Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively). This option allows updating stream structure so that standard consumer may detect the change. To do so, reopen output "AVFormatContext" (in case of API usage) or restart ffmpeg instance, cyclically changing tables_version value:
ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
...
ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
...

Example

ffmpeg -i file.mpg -c copy \
     -mpegts_original_network_id 0x1122 \
     -mpegts_transport_stream_id 0x3344 \
     -mpegts_service_id 0x5566 \
     -mpegts_pmt_start_pid 0x1500 \
     -mpegts_start_pid 0x150 \
     -metadata service_provider="Some provider" \
     -metadata service_name="Some Channel" \
     out.ts

MXF muxer.

Options

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 and mxf_opatom 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, -C<write_index> 0 can be used to disable
all growing data tables, allowing to mux endless streams with limited memory
and without these disadvantages.

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.

Options

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:

cache
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.

Examples

  • Remux the content of file in.mkv to a list of segments out-000.nut, out-001.nut, etc., and write the list of generated segments to out.list:
    ffmpeg -i in.mkv -codec hevc -flags +cgop -g 60 -map 0 -f segment -segment_list out.list out%03d.nut
    
  • Segment input and set output format options for the output segments:
    ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4
    
  • Segment the input file according to the split points specified by the segment_times option:
    ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut
    
  • Use the ffmpeg force_key_frames option to force key frames in the input at the specified location, together with the segment option segment_time_delta to account for possible roundings operated when setting key frame times.
    ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
    -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut
    

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

  • Segment the input file by splitting the input file according to the frame numbers sequence specified with the segment_frames option:
    ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut
    
  • Convert the in.mkv to TS segments using the "libx264" and "aac" encoders:
    ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts
    
  • Segment the input file, and create an M3U8 live playlist (can be used as live HLS source):
    ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
    -segment_list_flags +live -segment_time 10 out%03d.mkv
    

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).

Per stream hash testing format.

This muxer computes and prints a cryptographic hash of all the input frames, on a per-stream basis. 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 one line per stream of the form: streamindex,streamtype,algo=hash, where streamindex is the index of the mapped stream, streamtype is a single character indicating the type of stream, algo is a short string representing the hash function used, and hash is a hexadecimal number representing the computed hash.

hash algorithm
Use the cryptographic hash function specified by the string algorithm. Supported values include "MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256" (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32" and "adler32".

Examples

To compute the SHA-256 hash of the input converted to raw audio and video, and store it in the file out.sha256:

ffmpeg -i INPUT -f streamhash out.sha256

To print an MD5 hash to stdout use the command:

ffmpeg -i INPUT -f streamhash -hash md5 -

See also the hash and framehash muxers.

The fifo pseudo-muxer allows the separation of encoding and muxing by using first-in-first-out queue and running the actual muxer in a separate thread. This is especially useful in combination with the tee muxer and can be used to send data to several destinations with different reliability/writing speed/latency.

API users should be aware that callback functions (interrupt_callback, io_open and io_close) used within its AVFormatContext must be thread-safe.

The behavior of the fifo muxer if the queue fills up or if the output fails is selectable,

  • output can be transparently restarted with configurable delay between retries based on real time or time of the processed stream.
  • encoding can be blocked during temporary failure, or continue transparently dropping packets in case fifo queue fills up.
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.
Buffer the specified amount of packets and delay writing the output. Note that queue_size must be big enough to store the packets for timeshift. At the end of the input the fifo buffer is flushed at realtime speed.

Examples

Stream something to rtmp server, continue processing the stream at real-time rate even in case of temporary failure (network outage) and attempt to recover streaming every second indefinitely.
ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a
  -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name

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

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

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

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

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

Options

If set to 1, slave outputs will be processed in separate threads using the fifo muxer. This allows to compensate for different speed/latency/reliability of outputs and setup transparent recovery. By default this feature is turned off.
Options to pass to fifo pseudo-muxer instances. See fifo.

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

The following special options are also recognized:

Specify the format name. Required if it cannot be guessed from the output URL.
Specify a list of bitstream filters to apply to the specified output.

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

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

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

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

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.
Select the streams that should be mapped to the slave output, specified by a stream specifier. If not specified, this defaults to all the mapped streams. This will cause that output operation to fail if the output format does not accept all mapped streams.

You may use multiple stream specifiers separated by commas (",") e.g.: "a:0,v"

Specify behaviour on output failure. This can be set to either "abort" (which is default) or "ignore". "abort" will cause whole process to fail in case of failure on this slave output. "ignore" will ignore failure on this output, so other outputs will continue without being affected.

Examples

  • Encode something and both archive it in a WebM file and stream it as MPEG-TS over UDP:
    ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
      "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
    
  • As above, but continue streaming even if output to local file fails (for example local drive fills up):
    ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
      "[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
    
  • Use ffmpeg to encode the input, and send the output to three different destinations. The "dump_extra" bitstream filter is used to add extradata information to all the output video keyframes packets, as requested by the MPEG-TS format. The select option is applied to out.aac in order to make it contain only audio packets.
    ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
           -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
    
  • As above, but select only stream "a:1" for the audio output. Note that a second level escaping must be performed, as ":" is a special character used to separate options.
    ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
           -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
    

WebM 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:

Options

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.

Example

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

WebM Live Chunk Muxer.

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

Options

This muxer supports the following options:

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).

Example

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

FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded INI-like text file and then load it back using the metadata muxer/demuxer.

The file format is as follows:

1.
A file consists of a header and a number of metadata tags divided into sections, each on its own line.
2.
The header is a ;FFMETADATA string, followed by a version number (now 1).
3.
Metadata tags are of the form key=value
4.
Immediately after header follows global metadata
5.
After global metadata there may be sections with per-stream/per-chapter metadata.
6.
A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in brackets ([, ]) and ends with next section or end of file.
7.
At the beginning of a chapter section there may be an optional timebase to be used for start/end values. It must be in form TIMEBASE=num/den, where num and den are integers. If the timebase is missing then start/end times are assumed to be in nanoseconds.

Next a chapter section must contain chapter start and end times in form START=num, END=num, where num is a positive integer.

8.
Empty lines and lines starting with ; or # are ignored.
9.
Metadata keys or values containing special characters (=, ;, #, \ and a newline) must be escaped with a backslash \.
10.
Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of the tag (in the example above key is foo , value is
bar).

A ffmetadata file might look like this:

;FFMETADATA1
title=bike\\shed
;this is a comment
artist=FFmpeg troll team

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

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

Extracting an ffmetadata file with ffmpeg goes as follows:

ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

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

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

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.

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

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

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

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

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

Muliple subscribers may stream from the broker using the command:

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

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

The following options are supported:

Sets the exchange to use on the broker. RabbitMQ has several predefined exchanges: "amq.direct" is the default exchange, where the publisher and subscriber must have a matching routing_key; "amq.fanout" is the same as a broadcast operation (i.e. the data is forwarded to all queues on the fanout exchange independent of the routing_key); and "amq.topic" is similar to "amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ documentation).
Sets the routing key. The default value is "amqp". The routing key is used on the "amq.direct" and "amq.topic" exchanges to decide whether packets are written to the queue of a subscriber.
Maximum size of each packet sent/received to the broker. Default is 131072. Minimum is 4096 and max is any large value (representable by an int). When receiving packets, this sets an internal buffer size in FFmpeg. It should be equal to or greater than the size of the published packets to the broker. Otherwise the received message may be truncated causing decoding errors.
The timeout in seconds during the initial connection to the broker. The default value is rw_timeout, or 5 seconds if rw_timeout is not set.
Sets the delivery mode of each message sent to broker. The following values are accepted:
Delivery mode set to "persistent" (2). This is the default value. Messages may be written to the broker's disk depending on its setup.
Delivery mode set to "non-persistent" (1). Messages will stay in broker's memory unless the broker is under memory pressure.

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.

The accepted options are:

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

URL Syntax is

cache:<URL>

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

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

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

Gopher protocol.

Gophers protocol.

The Gopher protocol with TLS encapsulation.

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.
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.
Reconnect automatically in case of TCP/TLS errors during connect.
A comma separated list of HTTP status codes to reconnect on. The list can include specific status codes (e.g. '503') or the strings '4xx' / '5xx'.
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>
Send an Expect: 100-continue header for POST. If set to 1 it will send, if set to 0 it won't, if set to -1 it will try to send if it is applicable. Default value is -1.
Set HTTP authentication type. No option for Digest, since this method requires getting nonce parameters from the server first and can't be used straight away like Basic.
Choose the HTTP authentication type automatically. This is the default.
Choose the HTTP basic authentication.

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

HTTP Cookies

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

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

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

Icecast 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.
tls
Establish a TLS (HTTPS) connection to Icecast.
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>

Reliable Internet Streaming Transport protocol

The accepted options are:

Supported values:
This one is default.
Set internal RIST buffer size in milliseconds for retransmission of data. Default value is 0 which means the librist default (1 sec). Maximum value is 30 seconds.
Set maximum packet size for sending data. 1316 by default.
Set loglevel for RIST logging messages. You only need to set this if you explicitly want to enable debug level messages or packet loss simulation, otherwise the regular loglevel is respected.
Set override of encryption secret, by default is unset.
Set encryption type, by default is disabled. Acceptable values are 128 and 256.

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 "AVOption"s):

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.
Set the maximum UDP socket buffer size in bytes.
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.
Set timeout (in microseconds) of socket I/O operations to n.

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

Important notes:

1.
If rtcpport is not set the RTCP port will be set to the RTP port value plus 1.
2.
If localrtpport (the local RTP port) is not set any available port will be used for the local RTP and RTCP ports.
3.
If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port value plus 1.

Real-Time Streaming Protocol.

RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with data transferred over RDT).

The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's https://github.com/revmischa/rtsp-server).

The required syntax for a RTSP url is:

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

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

The following options are supported.

Do not start playing the stream immediately if set to 1. Default value is 0.
Set RTSP transport protocols.

It accepts the following values:

udp
Use UDP as lower transport protocol.
tcp
Use TCP (interleaving within the RTSP control channel) as lower transport protocol.
Use UDP multicast as lower transport protocol.
http
Use HTTP tunneling as lower transport protocol, which is useful for passing proxies.

Multiple lower transport protocols may be specified, in that case they are tried one at a time (if the setup of one fails, the next one is tried). For the muxer, only the tcp and udp options are supported.

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:

data

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".

Examples

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

  • Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
    ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
    
  • Watch a stream tunneled over HTTP:
    ffplay -rtsp_transport http rtsp://server/video.mp4
    
  • Send a stream in realtime to a RTSP server, for others to watch:
    ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
    
  • Receive a stream in realtime:
    ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>
    

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

Muxer

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

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

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

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]

Demuxer

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

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

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

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

Example command lines follow.

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

ffplay sap://

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

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

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.
Timestamp-based Packet Delivery Delay. Used to absorb bursts of missed packet retransmissions. This flag sets both rcvlatency and peerlatency to the same value. Note that prior to version 1.3.0 this is the only flag to set the latency, however this is effectively equivalent to setting peerlatency, when side is sender and rcvlatency when side is receiver, and the bidirectional stream sending is not supported.
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).
If true, both connection parties must have the same password set (including empty, that is, with no encryption). If the password doesn't match or only one side is unencrypted, the connection is rejected. Default is true.
The number of packets to be transmitted after which the encryption key is switched to a new key. Default is -1. -1 means auto (0x1000000 in srt library). The range for this option is integers in the 0 - "INT_MAX".
The interval between when a new encryption key is sent and when switchover occurs. This value also applies to the subsequent interval between when switchover occurs and when the old encryption key is decommissioned. Default is -1. -1 means auto (0x1000 in srt library). The range for this option is integers in the 0 - "INT_MAX".
Sets the maximum declared size of a packet transferred during the single call to the sending function in Live mode. Use 0 if this value isn't used (which is default in file mode). Default is -1 (automatic), which typically means MPEG-TS; if you are going to use SRT to send any different kind of payload, such as, for example, wrapping a live stream in very small frames, then you can use a bigger maximum frame size, though not greater than 1456 bytes.
Alias for payload_size.
The latency value (as described in rcvlatency) that is set by the sender side as a minimum value for the receiver.
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.
The time that should elapse since the moment when the packet was sent and the moment when it's delivered to the receiver application in the receiving function. This time should be a buffer time large enough to cover the time spent for sending, unexpectedly extended RTT time, and the time needed to retransmit the lost UDP packet. The effective latency value will be the maximum of this options' value and the value of peerlatency set by the peer side. Before version 1.3.0 this option is only available as latency.
Set UDP receive buffer size, expressed in bytes.
Set UDP send buffer size, expressed in bytes.
Set raise error timeouts for read, write and connect operations. Note that the SRT library has internal timeouts which can be controlled separately, the value set here is only a cap on those.
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.
Set send buffer size, expressed in bytes.
Set receive buffer size, expressed in bytes.

Receive buffer must not be greater than ffs.

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

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

A string limited to 512 characters that can be set on the socket prior to connecting. This stream ID will be able to be retrieved by the listener side from the socket that is returned from srt_accept and was connected by a socket with that set stream ID. SRT does not enforce any special interpretation of the contents of this string. This option doesnXt make sense in Rendezvous connection; the result might be that simply one side will override the value from the other side and itXs the matter of luck which one would win
The type of Smoother used for the transmission for that socket, which is responsible for the transmission and congestion control. The Smoother type must be exactly the same on both connecting parties, otherwise the connection is rejected.
When set, this socket uses the Message API, otherwise it uses Buffer API. Note that in live mode (see transtype) thereXs only message API available. In File mode you can chose to use one of two modes:

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

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

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

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

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

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

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. 0 disables listen, 1 enables listen in single client mode, 2 enables listen in multi-client mode. 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.
Set maximum segment size for outgoing TCP packets, expressed in bytes.

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

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

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

The required syntax for a TLS/SSL url is:

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

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

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

Example command lines:

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

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

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

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

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 32 KB for output, 384 KB for input. 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.
Local IP address of a network interface used for sending packets or joining multicast groups.
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 from the specified addresses. In case of multicast, also subscribe to multicast traffic coming from these addresses only.
Ignore packets sent from the specified addresses. In case of multicast, also exclude the source addresses in the multicast subscription.
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.

Examples

  • Use ffmpeg to stream over UDP to a remote endpoint:
    ffmpeg -i <input> -f <format> udp://<hostname>:<port>
    
  • Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
    ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535
    
  • Use ffmpeg to receive over UDP from a remote endpoint:
    ffmpeg -i udp://[<multicast-address>]:<port> ...
    

Unix local socket

The required syntax for a Unix socket URL is:

unix://<filepath>

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

Timeout in ms.
Create the Unix socket in listening mode.

ZeroMQ asynchronous messaging using the libzmq library.

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

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

zmq:tcp://ip-address:port

Example: Create a localhost stream on port 5555:

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

Multiple clients may connect to the stream using:

ffplay zmq:tcp://127.0.0.1:5555

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

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

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

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

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

Options

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.

Options

Set the video size given as a string such as 640x480 or hd720. Falls back to the first available configuration reported by Android if requested video size is not available or by default.
framerate
Set the video framerate. Falls back to the first available configuration reported by Android if requested framerate is not available or by default (-1).
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

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

and/or

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

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

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

There are two device name aliases:

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

Options

AVFoundation supports the following options:

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

Examples

  • Print the list of AVFoundation supported devices and exit:
    $ ffmpeg -f avfoundation -list_devices true -i ""
    
  • Record video from video device 0 and audio from audio device 0 into out.avi:
    $ ffmpeg -f avfoundation -i "0:0" out.avi
    
  • Record video from video device 2 and audio from audio device 1 into out.avi:
    $ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
    
  • Record video from the system default video device using the pixel format bgr0 and do not record any audio into out.avi:
    $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
    
  • Record raw DV data from a suitable input device and write the output into out.dv:
    $ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv
    

BSD video input device.

Options

framerate
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.

Options

If set to true, print a list of devices and exit. Defaults to false. This option is deprecated, please use the "-sources" option of ffmpeg to list the available input devices.
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.
Set the pixel format of the captured video. Available values are:
This is the default which means 8-bit YUV 422 or 8-bit ARGB if format autodetection is used, 8-bit YUV 422 otherwise.
8-bit YUV 422.
10-bit YUV 422.
8-bit RGB.
8-bit RGB.
10-bit RGB.
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.
Timecode type to include in the frame and video stream metadata. Must be none, rp188vitc, rp188vitc2, rp188ltc, rp188hfr, rp188any, vitc, vitc2, or serial. Defaults to none (not included).

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

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.
Capture start time alignment in seconds. If set to nonzero, input frames are dropped till the system timestamp aligns with configured value. Alignment difference of up to one frame duration is tolerated. This is useful for maintaining input synchronization across N different hardware devices deployed for 'N-way' redundancy. The system time of different hardware devices should be synchronized with protocols such as NTP or PTP, before using this option. Note that this method is not foolproof. In some border cases input synchronization may not happen due to thread scheduling jitters in the OS. Either sync could go wrong by 1 frame or in a rarer case timestamp_align seconds. Defaults to 0.
Drop frames till a frame with timecode is received. Sometimes serial timecode isn't received with the first input frame. If that happens, the stored stream timecode will be inaccurate. If this option is set to true, input frames are dropped till a frame with timecode is received. Option timecode_format must be specified. Defaults to false.
If set to true, extracts KLV data from VANC and outputs KLV packets. KLV VANC packets are joined based on MID and PSC fields and aggregated into one KLV packet. Defaults to false.

Examples

  • List input devices:
    ffmpeg -sources decklink
    
  • List supported formats:
    ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
    
  • Capture video clip at 1080i50:
    ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi
    
  • Capture video clip at 1080i50 10 bit:
    ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
    
  • Capture video clip at 1080i50 with 16 audio channels:
    ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
    

Windows DirectShow input device.

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

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

The input name should be in the format:

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

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

Options

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

Set the video size in the captured video.
framerate
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.

Examples

  • Print the list of DirectShow supported devices and exit:
    $ ffmpeg -list_devices true -f dshow -i dummy
    
  • Open video device Camera:
    $ ffmpeg -f dshow -i video="Camera"
    
  • Open second video device with name Camera:
    $ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
    
  • Open video device Camera and audio device Microphone:
    $ ffmpeg -f dshow -i video="Camera":audio="Microphone"
    
  • Print the list of supported options in selected device and exit:
    $ ffmpeg -list_options true -f dshow -i video="Camera"
    
  • Specify pin names to capture by name or alternative name, specify alternative device name:
    $ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"
    
  • Configure a crossbar device, specifying crossbar pins, allow user to adjust video capture properties at startup:
    $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
         -crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"
    

Linux framebuffer input device.

The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on a computer monitor, typically on the console. It is accessed through a file device node, usually /dev/fb0.

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

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

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

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

You can take a single screenshot image with the command:

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

Options

framerate
Set the frame rate. Default is 25.

Win32 GDI-based screen capture device.

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

There are two options for the input filename:

desktop

or

title=<window_title>

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

For example, to grab the entire desktop using ffmpeg:

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

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

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

Grab the contents of the window named "Calculator"

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

Options

Specify whether to draw the mouse pointer. Use the value 0 to not draw the pointer. Default value is 1.
framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of "30000/1001".
Show 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.

Options

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.

Examples

  • Grab and show the input of a FireWire DV/HDV device.
    ffplay -f iec61883 -i auto
    
  • Grab and record the input of a FireWire DV/HDV device, using a packet buffer of 100000 packets if the source is HDV.
    ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg
    

JACK input device.

To enable this input device during configuration you need libjack installed on your system.

A JACK input device creates one or more JACK writable clients, one for each audio channel, with name client_name:input_N, where client_name is the name provided by the application, and N is a number which identifies the channel. Each writable client will send the acquired data to the FFmpeg input device.

Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients.

To connect or disconnect JACK clients you can use the jack_connect and jack_disconnect programs, or do it through a graphical interface, for example with qjackctl.

To list the JACK clients and their properties you can invoke the command jack_lsp.

Follows an example which shows how to capture a JACK readable client with ffmpeg.

# Create a JACK writable client with name "ffmpeg".
$ ffmpeg -f jack -i ffmpeg -y out.wav

# Start the sample jack_metro readable client.
$ jack_metro -b 120 -d 0.2 -f 4000

# List the current JACK clients.
$ jack_lsp -c
system:capture_1
system:capture_2
system:playback_1
system:playback_2
ffmpeg:input_1
metro:120_bpm

# Connect metro to the ffmpeg writable client.
$ jack_connect metro:120_bpm ffmpeg:input_1

For more information read: http://jackaudio.org/

Options

Set the number of channels. Default is 2.

KMS video input device.

Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a DRM object that can be passed to other hardware functions.

Requires either DRM master or CAP_SYS_ADMIN to run.

If you don't understand what all of that means, you probably don't want this. Look at x11grab instead.

Options

DRM device to capture on. Defaults to /dev/dri/card0.
format
Pixel format of the framebuffer. This can be autodetected if you are running Linux 5.7 or later, but needs to be provided for earlier versions. Defaults to bgr0, which is the most common format used by the Linux console and Xorg X server.
Format modifier to signal on output frames. This is necessary to import correctly into some APIs. It can be autodetected if you are running Linux 5.7 or later, but will need to be provided explicitly when needed in earlier versions. See the libdrm documentation for possible values.
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
Framerate to capture at. This is not synchronised to any page flipping or framebuffer changes - it just defines the interval at which the framebuffer is sampled. Sampling faster than the framebuffer update rate will generate independent frames with the same content. Defaults to 30.

Examples

  • Capture from the first active plane, download the result to normal frames and encode. This will only work if the framebuffer is both linear and mappable - if not, the result may be scrambled or fail to download.
    ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
    
  • Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
    ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
    
  • To capture only part of a plane the output can be cropped - this can be used to capture a single window, as long as it has a known absolute position and size. For example, to capture and encode the middle quarter of a 1920x1080 plane:
    ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4
    

Libavfilter input virtual device.

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

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

Options

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.

Examples

  • Create a color video stream and play it back with ffplay:
    ffplay -f lavfi -graph "color=c=pink [out0]" dummy
    
  • As the previous example, but use filename for specifying the graph description, and omit the "out0" label:
    ffplay -f lavfi color=c=pink
    
  • Create three different video test filtered sources and play them:
    ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
    
  • Read an audio stream from a file using the amovie source and play it back with ffplay:
    ffplay -f lavfi "amovie=test.wav"
    
  • Read an audio stream and a video stream and play it back with ffplay:
    ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
    
  • Dump decoded frames to images and closed captions to a file (experimental):
    ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin
    

Audio-CD input device based on libcdio.

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

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

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

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

Options

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".

Options

framerate
Set the frame rate. Default is "ntsc", corresponding to a frame rate of "30000/1001".
Select the pixel format. Default is "uyvy422".
Set the video size given as a string such as "640x480" or "hd720". Default is "qvga".

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.

Options

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.

Examples

Print the list of OpenAL supported devices and exit:

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

Capture from the OpenAL device DR-BT101 via PulseAudio:

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

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

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

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

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

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

Open Sound System input device.

The filename to provide to the input device is the device node representing the OSS input device, and is usually set to /dev/dsp.

For example to grab from /dev/dsp using ffmpeg use the command:

ffmpeg -f oss -i /dev/dsp /tmp/oss.wav

For more information about OSS see: http://manuals.opensound.com/usersguide/dsp.html

Options

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.

Options

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.

Examples

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

Options

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:

  • List supported formats for a video4linux2 device:
    ffplay -f video4linux2 -list_formats all /dev/video0
    
  • Grab and show the input of a video4linux2 device:
    ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
    
  • Grab and record the input of a video4linux2 device, leave the frame rate and size as previously set:
    ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
    

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

Options

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.
framerate
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.

Options

Set the video frame size.
framerate
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

Options

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

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

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

This option disables options follow_mouse and select_region.

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

Examples

  • Play a file on default ALSA device:
    ffmpeg -i INPUT -f alsa default
    
  • Play a file on soundcard 1, audio device 7:
    ffmpeg -i INPUT -f alsa hw:1,7
    

AudioToolbox output device.

Allows native output to CoreAudio devices on OSX.

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

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

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

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

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

Options

AudioToolbox supports the following options:

Specify the audio device by its index. Overrides anything given in the output filename.

Examples

  • Print the list of supported devices and output a sine wave to the default device:
    $ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -list_devices true -
    
  • Output a sine wave to the device with the index 2, overriding any output filename:
    $ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -audio_device_index 2 -
    

CACA output device.

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

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

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

Options

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".

Examples

  • The following command shows the ffmpeg output is an CACA window, forcing its size to 80x25:
    ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -
    
  • Show the list of available drivers and exit:
    ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -
    
  • Show the list of available dither colors and exit:
    ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
    

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

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

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

Options

If set to true, print a list of devices and exit. Defaults to false. This option is deprecated, please use the "-sinks" option of ffmpeg to list the available output devices.
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.
Sets the decklink device duplex mode. Must be unset, half or full. Defaults to unset.
Sets the genlock timing pixel offset on the used output. Defaults to unset.

Examples

  • List output devices:
    ffmpeg -sinks decklink
    
  • List supported formats:
    ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
    
  • Play video clip:
    ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
    
  • Play video clip with non-standard framerate or video size:
    ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
    

Linux framebuffer output device.

The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on a computer monitor, typically on the console. It is accessed through a file device node, usually /dev/fb0.

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

Options

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

Examples

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

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

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

OpenGL output device.

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

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

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

Options

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.

Examples

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

Options

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.

Examples

Play a file on default device on default server:

ffmpeg  -i INPUT -f pulse "stream name"

SDL (Simple DirectMedia Layer) output device.

"sdl2" can be used as alias for "sdl".

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

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

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

Options

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 the position of the window on the screen.
Set fullscreen mode when non-zero value is provided. Default value is zero.
Enable quit action (using window button or keyboard key) when non-zero value is provided. Default value is 1 (enable quit action)

Interactive commands

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

Quit the device immediately.

Examples

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

ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"

sndio audio output device.

Video4Linux2 output device.

XV (XVideo) output device.

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

Options

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/.

Examples

  • Decode, display and encode video input with ffmpeg at the same time:
    ffmpeg -i INPUT OUTPUT -f xv display
    
  • Decode and display the input video to multiple X11 windows:
    ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
    

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 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.
async
For swr only, simple 1 parameter audio sync to timestamps using stretching, squeezing, filling and trimming. Setting this to 1 will enable filling and trimming, larger values represent the maximum amount in samples that the data may be stretched or squeezed for each second. Default value is 0, thus no compensation is applied to make the samples match the audio timestamps.
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, with a few API-only exceptions noted below. For programmatic use, they can be set explicitly in the "SwsContext" options or through the libavutil/opt.h API.

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.
sinc
Select sinc rescaling algorithm.
Select Lanczos rescaling algorithm. The default width (alpha) is 3 and can be changed by setting "param0".
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).
If value is set to 1, indicates source is full range. Default value is 0, which indicates source is limited range.
If value is set to 1, enable full range for destination. Default value is 0, which enables limited 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:

  • A ':'-separated list of key=value pairs.
  • A ':'-separated list of value. In this case, the keys are assumed to be the option names in the order they are declared. E.g. the "fade" filter declares three options in this order -- type, start_frame and nb_frames. Then the parameter list in:0:30 means that the value in is assigned to the option type, 0 to start_frame and 30 to nb_frames.
  • A ':'-separated list of mixed direct value and long key=value pairs. The direct value must precede the key=value pairs, and follow the same constraints order of the previous point. The following key=value pairs can be set in any preferred order.

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

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

The name and arguments of the filter are optionally preceded and followed by a list of link labels. A link label allows one to name a link and associate it to a filter output or input pad. The preceding labels in_link_1 ... in_link_N, are associated to the filter input pads, the following labels out_link_1 ... out_link_M, are associated to the output pads.

When two link labels with the same name are found in the filtergraph, a link between the corresponding input and output pad is created.

If an output pad is not labelled, it is linked by default to the first unlabelled input pad of the next filter in the filterchain. For example in the filterchain

nullsrc, split[L1], [L2]overlay, nullsink

the split filter instance has two output pads, and the overlay filter instance two input pads. The first output pad of split is labelled "L1", the first input pad of overlay is labelled "L2", and the second output pad of split is linked to the second input pad of overlay, which are both unlabelled.

In a filter description, if the input label of the first filter is not specified, "in" is assumed; if the output label of the last filter is not specified, "out" is assumed.

In a complete filterchain all the unlabelled filter input and output pads must be connected. A filtergraph is considered valid if all the filter input and output pads of all the filterchains are connected.

Libavfilter will automatically insert scale filters where format conversion is required. It is possible to specify swscale flags for those automatically inserted scalers by prepending "sws_flags=flags;" to the filtergraph description.

Here is a BNF description of the filtergraph syntax:

<NAME>             ::= sequence of alphanumeric characters and '_'
<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 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 options can be changed during the operation of the filter using a command. These options are marked 'T' on the output of ffmpeg -h filter=<name of filter>. The name of the command is the name of the option and the argument is the new value.

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.
Set mode of compressor operation. Can be "upward" or "downward". Default is "downward".
threshold
If a signal of stream rises above this level it will affect the gain reduction. By default it is 0.125. Range is between 0.00097563 and 1.
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.
mix
How much to use compressed signal in output. Default is 1. Range is between 0 and 1.

Commands

This filter supports the all above options as commands.

Simple audio dynamic range compression/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 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.

Examples

  • Cross fade from one input to another:
    ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac
    
  • Cross fade from one input to another but without overlapping:
    ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac
    

Split audio stream into several bands.

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

The filter accepts the following options:

Set split frequencies. Those must be positive and increasing.
Set filter order for each band split. This controls filter roll-off or steepness of filter transfer function. Available values are:
2nd
12 dB per octave.
4th
24 dB per octave.
6th
36 dB per octave.
8th
48 dB per octave.
10th
60 dB per octave.
12th
72 dB per octave.
14th
84 dB per octave.
16th
96 dB per octave.
18th
108 dB per octave.
20th
120 dB per octave.

Default is 4th.

Set input gain level. Allowed range is from 0 to 1. Default value is 1.
Set output gain for each band. Default value is 1 for all bands.

Examples

  • Split input audio stream into two bands (low and high) with split frequency of 1500 Hz, each band will be in separate stream:
    ffmpeg -i in.flac -filter_complex 'acrossover=split=1500[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav
    
  • Same as above, but with higher filter order:
    ffmpeg -i in.flac -filter_complex 'acrossover=split=1500:order=8th[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav
    
  • Same as above, but also with additional middle band (frequencies between 1500 and 8000):
    ffmpeg -i in.flac -filter_complex 'acrossover=split=1500 8000:order=8th[LOW][MID][HIGH]' -map '[LOW]' low.wav -map '[MID]' mid.wav -map '[HIGH]' high.wav
    

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.
mix
Set mixing amount.
Can be linear: "lin" or logarithmic: "log".
Set DC.
aa
Set anti-aliasing.
Set sample reduction.
Enable LFO. By default disabled.
Set LFO range.
Set LFO rate.

Commands

This filter supports the all above options as commands.

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

Remove impulsive noise from input audio.

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

Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55 milliseconds. This sets size of window which will be processed at once.
Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default value is 75 percent. Setting this to a very high value increases impulsive noise removal but makes whole process much slower.
Set autoregression order, in percentage of window size. Allowed range is from 0 to 25. Default value is 2 percent. This option also controls quality of interpolated samples using neighbour good samples.
Set threshold value. Allowed range is from 1 to 100. Default value is 2. This controls the strength of impulsive noise which is going to be removed. The lower value, the more samples will be detected as impulsive noise.
Set burst fusion, in percentage of window size. Allowed range is 0 to 10. Default value is 2. If any two samples detected as noise are spaced less than this value then any sample between those two samples will be also detected as noise.
Set overlap method.

It accepts the following values:

Select overlap-add method. Even not interpolated samples are slightly changed with this method.
Select overlap-save method. Not interpolated samples remain unchanged.

Default value is "a".

Remove clipped samples from input audio.

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

Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55 milliseconds. This sets size of window which will be processed at once.
Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default value is 75 percent.
Set autoregression order, in percentage of window size. Allowed range is from 0 to 25. Default value is 8 percent. This option also controls quality of interpolated samples using neighbour good samples.
Set threshold value. Allowed range is from 1 to 100. Default value is 10. Higher values make clip detection less aggressive.
Set size of histogram used to detect clips. Allowed range is from 100 to 9999. Default value is 1000. Higher values make clip detection less aggressive.
Set overlap method.

It accepts the following values:

Select overlap-add method. Even not interpolated samples are slightly changed with this method.
Select overlap-save method. Not interpolated samples remain unchanged.

Default value is "a".

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. If you want instead to delay in seconds, append 's' to number.
Use last set delay for all remaining channels. By default is disabled. This option if enabled changes how option "delays" is interpreted.

Examples

  • Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave the second channel (and any other channels that may be present) unchanged.
    adelay=1500|0|500
    
  • Delay second channel by 500 samples, the third channel by 700 samples and leave the first channel (and any other channels that may be present) unchanged.
    adelay=0|500S|700S
    
  • Delay all channels by same number of samples:
    adelay=delays=64S:all=1
    

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

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

A description of the accepted parameters follows.

Set level of added noise in dB. Default is "-351". Allowed range is from -451 to -90.
Set type of added noise.
Add DC signal.
Add AC signal.
Add square signal.
pulse
Add pulse signal.

Default is "dc".

Commands

This filter supports the all above options as commands.

Compute derivative/integral of audio stream.

Applying both filters one after another produces original audio.

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.

Examples

  • Make it sound as if there are twice as many instruments as are actually playing:
    aecho=0.8:0.88:60:0.4
    
  • If delay is very short, then it sounds like a (metallic) robot playing music:
    aecho=0.8:0.88:6:0.4
    
  • A longer delay will sound like an open air concert in the mountains:
    aecho=0.8:0.9:1000:0.3
    
  • Same as above but with one more mountain:
    aecho=0.8:0.9:1000|1800:0.3|0.25
    

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).
50fm
select 50Xs (FM).
75fm
select 75Xs (FM).
50kf
select 50Xs (FM-KF).
75kf
select 75Xs (FM-KF).

Commands

This filter supports the all above options as commands.

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.

Examples

  • Half volume:
    aeval=val(ch)/2:c=same
    
  • Invert phase of the second channel:
    aeval=val(0)|-val(1)
    

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

The filter accepts the following options:

Set input level prior processing of signal. Allowed range is from 0 to 64. Default value is 1.
Set output level after processing of signal. Allowed range is from 0 to 64. Default value is 1.
Set the amount of harmonics added to original signal. Allowed range is from 0 to 64. Default value is 1.
Set the amount of newly created harmonics. Allowed range is from 0.1 to 10. Default value is 8.5.
blend
Set the octave of newly created harmonics. Allowed range is from -10 to 10. Default value is 0.
Set the lower frequency limit of producing harmonics in Hz. Allowed range is from 2000 to 12000 Hz. Default is 7500 Hz.
Set the upper frequency limit of producing harmonics. Allowed range is from 9999 to 20000 Hz. If value is lower than 10000 Hz no limit is applied.
Mute the original signal and output only added harmonics. By default is disabled.

Commands

This filter supports the all above options as commands.

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 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 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
select logistic sigmoid
sinc
select sine cardinal function
select inverted sine cardinal function
no fade applied

Commands

This filter supports the all above options as commands.

Examples

  • Fade in first 15 seconds of audio:
    afade=t=in:ss=0:d=15
    
  • Fade out last 25 seconds of a 900 seconds audio:
    afade=t=out:st=875:d=25
    

Denoise audio samples with FFT.

A description of the accepted parameters follows.

Set the noise reduction in dB, allowed range is 0.01 to 97. Default value is 12 dB.
Set the noise floor in dB, allowed range is -80 to -20. Default value is -50 dB.
Set the noise type.

It accepts the following values:

Select white noise.
Select vinyl noise.
Select shellac noise.
Select custom noise, defined in "bn" option.

Default value is white noise.

Set custom band noise for every one of 15 bands. Bands are separated by ' ' or '|'.
Set the residual floor in dB, allowed range is -80 to -20. Default value is -38 dB.
Enable noise tracking. By default is disabled. With this enabled, noise floor is automatically adjusted.
Enable residual tracking. By default is disabled.
Set the output mode.

It accepts the following values:

Pass input unchanged.
Pass noise filtered out.
Pass only noise.

Default value is o.

Commands

This filter supports the following commands:

Start or stop measuring noise profile. Syntax for the command is : "start" or "stop" string. After measuring noise profile is stopped it will be automatically applied in filtering.
Change noise reduction. Argument is single float number. Syntax for the command is : "noise_reduction"
Change noise floor. Argument is single float number. Syntax for the command is : "noise_floor"
Change output mode operation. Syntax for the command is : "i", "o" or "n" string.

Apply arbitrary expressions to samples in frequency domain.

Set frequency domain real expression for each separate channel separated by '|'. Default is "re". If the number of input channels is greater than the number of expressions, the last specified expression is used for the remaining output channels.
Set frequency domain imaginary expression for each separate channel separated by '|'. Default is "im".

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

sr
sample rate
current frequency bin number
number of available bins
channel number of the current expression
number of channels
current frame pts
current real part of frequency bin of current channel
current imaginary part of frequency bin of current channel
Return the value of real part of frequency bin at location (bin,channel)
Return the value of imaginary part of frequency bin at location (bin,channel)
Set window size. Allowed range is from 16 to 131072. Default is 4096
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.

Examples

  • Leave almost only low frequencies in audio:
    afftfilt="'real=re * (1-clip((b/nb)*b,0,1))':imag='im * (1-clip((b/nb)*b,0,1))'"
    
  • Apply robotize effect:
    afftfilt="real='hypot(re,im)*sin(0)':imag='hypot(re,im)*cos(0)':win_size=512:overlap=0.75"
    
  • Apply whisper effect:
    afftfilt="real='hypot(re,im)*cos((random(0)*2-1)*2*3.14)':imag='hypot(re,im)*sin((random(1)*2-1)*2*3.14)':win_size=128:overlap=0.8"
    

Apply an arbitrary Finite Impulse Response filter.

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

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

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

It accepts the following parameters:

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.

Set which approach to use for auto gain measurement.

Do not apply any gain.
select peak gain, very conservative approach. This is default value.
select DC gain, limited application.
select gain to noise approach, this is most popular one.
Set gain to be applied to IR coefficients before filtering. Allowed range is 0 to 1. This gain is applied after any gain applied with gtype option.
Set format of IR stream. Can be "mono" or "input". Default is "input".
Set max allowed Impulse Response filter duration in seconds. Default is 30 seconds. Allowed range is 0.1 to 60 seconds.
Show IR frequency response, magnitude(magenta), phase(green) and group delay(yellow) in additional video stream. By default it is disabled.
Set for which IR channel to display frequency response. By default is first channel displayed. This option is used only when response is enabled.
Set video stream size. This option is used only when response is enabled.
Set video stream frame rate. This option is used only when response is enabled.
Set minimal partition size used for convolution. Default is 8192. Allowed range is from 1 to 32768. Lower values decreases latency at cost of higher CPU usage.
Set maximal partition size used for convolution. Default is 8192. Allowed range is from 8 to 32768. Lower values may increase CPU usage.
Set number of input impulse responses streams which will be switchable at runtime. Allowed range is from 1 to 32. Default is 1.
Set IR stream which will be used for convolution, starting from 0, should always be lower than supplied value by "nbirs" option. Default is 0. This option can be changed at runtime via commands.

Examples

Apply reverb to stream using mono IR file as second input, complete command using ffmpeg:
ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav

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 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

Apply frequency shift to input audio samples.

The filter accepts the following options:

Specify frequency shift. Allowed range is -INT_MAX to INT_MAX. Default value is 0.0.
Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value is 1.0.

Commands

This filter supports the all above options as commands.

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 mode of operation. Can be "upward" or "downward". Default is "downward". If set to "upward" mode, higher parts of signal will be amplified, expanding dynamic range in upward direction. Otherwise, in case of "downward" lower parts of signal will be reduced.
Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1. Setting this to 0 disables reduction and then filter behaves like expander.
threshold
If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.
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".

Commands

This filter supports the all above options as commands.

Apply an arbitrary Infinite Impulse Response filter.

It accepts the following parameters:

Set B/numerator/zeros/reflection coefficients.
Set A/denominator/poles/ladder coefficients.
Set channels gains.
Set input gain.
Set output gain.
Set coefficients format.
lattice-ladder function
analog transfer function
digital transfer function
Z-plane zeros/poles, cartesian (default)
Z-plane zeros/poles, polar radians
Z-plane zeros/poles, polar degrees
S-plane zeros/poles
Set type of processing.
direct processing
serial processing
parallel processing
Set filtering precision.
double-precision floating-point (default)
single-precision floating-point
32-bit integers
16-bit integers
Normalize filter coefficients, by default is enabled. Enabling it will normalize magnitude response at DC to 0dB.
mix
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Show IR frequency response, magnitude(magenta), phase(green) and group delay(yellow) in additional video stream. By default it is disabled.
Set for which IR channel to display frequency response. By default is first channel displayed. This option is used only when response is enabled.
Set video stream size. This option is used only when response is enabled.

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

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

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

Examples

  • Apply 2 pole elliptic notch at around 5000Hz for 48000 Hz sample rate:
    aiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf:r=d
    
  • Same as above but in "zp" format:
    aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s
    
  • Apply 3-rd order analog normalized Butterworth low-pass filter, using analog transfer function format:
    aiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d
    

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.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set the filter order, can be 1 or 2. Default is 2.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

Commands

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"
Change allpass mix. Syntax for the command is : "mix"

Loop audio samples.

The filter accepts the following options:

loop
Set the number of loops. Setting this value to -1 will result in infinite loops. Default is 0.
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.

Examples

  • Merge two mono files into a stereo stream:
    amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
    
  • Multiple merges assuming 1 video stream and 6 audio streams in input.mkv:
    ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv
    

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.
normalize
Always scale inputs instead of only doing summation of samples. Beware of heavy clipping if inputs are not normalized prior or after filtering by this filter if this option is disabled. By default is enabled.

Commands

This filter supports the following commands:

Syntax is same as option with same name.

Multiply first audio stream with second audio stream and store result in output audio stream. Multiplication is done by multiplying each sample from first stream with sample at same position from second stream.

With this element-wise multiplication one can create amplitude fades and amplitude modulations.

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:
0
Butterworth, this is default.
1
Chebyshev type 1.
2
Chebyshev type 2.
curves
With this option activated frequency response of anequalizer is displayed in video stream.
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.

Examples

Lower gain by 10 of central frequency 200Hz and width 100 Hz for first 2 channels using Chebyshev type 1 filter:
anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1

Commands

This filter supports the following commands:

Alter existing filter parameters. Syntax for the commands is : "fN|f=freq|w=width|g=gain"

fN is existing filter number, starting from 0, if no such filter is available error is returned. freq set new frequency parameter. width set new width parameter in Hertz. gain set new gain parameter in dB.

Full filter invocation with asendcmd may look like this: asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...

Reduce broadband noise in audio samples using Non-Local Means algorithm.

Each sample is adjusted by looking for other samples with similar contexts. This context similarity is defined by comparing their surrounding patches of size p. Patches are searched in an area of r around the sample.

The filter accepts the following options:

Set denoising strength. Allowed range is from 0.00001 to 10. Default value is 0.00001.
Set patch radius duration. Allowed range is from 1 to 100 milliseconds. Default value is 2 milliseconds.
Set research radius duration. Allowed range is from 2 to 300 milliseconds. Default value is 6 milliseconds.
Set the output mode.

It accepts the following values:

Pass input unchanged.
Pass noise filtered out.
Pass only noise.

Default value is o.

Set smooth factor. Default value is 11. Allowed range is from 1 to 15.

Commands

This filter supports the all above options as commands.

Apply Normalized Least-Mean-Squares algorithm to the first audio stream using the second audio stream.

This adaptive filter is used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean square of the error signal (difference between the desired, 2nd input audio stream and the actual signal, the 1st input audio stream).

A description of the accepted options follows.

Set filter order.
Set filter mu.
Set the filter eps.
Set the filter leakage.
It accepts the following values:
Pass the 1st input.
Pass the 2nd input.
Pass filtered samples.
Pass difference between desired and filtered samples.

Default value is o.

Examples

One of many usages of this filter is noise reduction, input audio is filtered with same samples that are delayed by fixed amount, one such example for stereo audio is:
asplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:out_mode=o

Commands

This filter supports the same commands as options, excluding option "order".

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.
Specify the duration of samples of silence to add. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-zero value.
Specify the minimum total duration in the output audio stream. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-zero value. If the value is longer than the input audio length, silence is added to the end, until the value is reached. This option is mutually exclusive with pad_dur

If neither the pad_len nor the whole_len nor pad_dur nor whole_dur option is set, the filter will add silence to the end of the input stream indefinitely.

Examples

  • Add 1024 samples of silence to the end of the input:
    apad=pad_len=1024
    
  • Make sure the audio output will contain at least 10000 samples, pad the input with silence if required:
    apad=whole_len=10000
    
  • Use ffmpeg to pad the audio input with silence, so that the video stream will always result the shortest and will be converted until the end in the output file when using the shortest option:
    ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT
    

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:

Apply phase shift to input audio samples.

The filter accepts the following options:

Specify phase shift. Allowed range is from -1.0 to 1.0. Default value is 0.0.
Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value is 1.0.

Commands

This filter supports the all above options as commands.

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 "Resampler Options" section in the ffmpeg-resampler(1) manual for the complete list of supported options.

Examples

  • Resample the input audio to 44100Hz:
    aresample=44100
    
  • Stretch/squeeze samples to the given timestamps, with a maximum of 1000 samples per second compensation:
    aresample=async=1000
    

Reverse an audio clip.

Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.

Examples

Take the first 5 seconds of a clip, and reverse it.
atrim=end=5,areverse

Reduce noise from speech using Recurrent Neural Networks.

This filter accepts the following options:

Set train model file to load. This option is always required.
mix
Set how much to mix filtered samples into final output. Allowed range is from -1 to 1. Default value is 1. Negative values are special, they set how much to keep filtered noise in the final filter output. Set this option to -1 to hear actual noise removed from input signal.

Commands

This filter supports the all above options as commands.

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.

Apply audio soft clipping.

Soft clipping is a type of distortion effect where the amplitude of a signal is saturated along a smooth curve, rather than the abrupt shape of hard-clipping.

This filter accepts the following options:

Set type of soft-clipping.

It accepts the following values:

threshold
Set threshold from where to start clipping. Default value is 0dB or 1.
Set gain applied to output. Default value is 0dB or 1.
Set additional parameter which controls sigmoid function.
Set oversampling factor.

Commands

This filter supports the all above options as commands.

Automatic Speech Recognition

This filter uses PocketSphinx for speech recognition. To enable compilation of this filter, you need to configure FFmpeg with "--enable-pocketsphinx".

It accepts the following options:

Set sampling rate of input audio. Defaults is 16000. This need to match speech models, otherwise one will get poor results.
Set dictionary containing acoustic model files.
Set pronunciation dictionary.
Set language model file.
Set language model set.
Set which language model to use.
Set output for log messages.

The filter exports recognized speech as the frame metadata "lavfi.asr.text".

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 Noise_floor Noise_floor_count Bit_depth Dynamic_range Zero_crossings Zero_crossings_rate Number_of_NaNs Number_of_Infs Number_of_denormals

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 Noise_floor Noise_floor_count Bit_depth Number_of_samples Number_of_NaNs Number_of_Infs Number_of_denormals

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.
Select the entries which need to be measured per channel. The metadata keys can be used as flags, default is all which measures everything. none disables all per channel measurement.
Select the entries which need to be measured overall. The metadata keys can be used as flags, default is all which measures everything. none disables all overall measurement.

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.
Minimum local peak measured in dBFS over a short window.
Number of occasions (not the number of samples) that the signal attained Noise floor.
Overall bit depth of audio. Number of bits used for each sample.
Measured dynamic range of audio in dB.
Number of points where the waveform crosses the zero level axis.
Rate of Zero crossings and number of audio samples.

Boost subwoofer frequencies.

The filter accepts the following options:

Set dry gain, how much of original signal is kept. Allowed range is from 0 to 1. Default value is 0.7.
Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to 1. Default value is 0.7.
Set delay line decay gain value. Allowed range is from 0 to 1. Default value is 0.7.
Set delay line feedback gain value. Allowed range is from 0 to 1. Default value is 0.9.
Set cutoff frequency in Hertz. Allowed range is 50 to 900. Default value is 100.
Set slope amount for cutoff frequency. Allowed range is 0.0001 to 1. Default value is 0.5.
Set delay. Allowed range is from 1 to 100. Default value is 20.

Commands

This filter supports the all above options as commands.

Cut subwoofer frequencies.

This filter allows to set custom, steeper roll off than highpass filter, and thus is able to more attenuate frequency content in stop-band.

The filter accepts the following options:

Set cutoff frequency in Hertz. Allowed range is 2 to 200. Default value is 20.
Set filter order. Available values are from 3 to 20. Default value is 10.
Set input gain level. Allowed range is from 0 to 1. Default value is 1.

Commands

This filter supports the all above options as commands.

Cut super frequencies.

The filter accepts the following options:

Set cutoff frequency in Hertz. Allowed range is 20000 to 192000. Default value is 20000.
Set filter order. Available values are from 3 to 20. Default value is 10.
Set input gain level. Allowed range is from 0 to 1. Default value is 1.

Commands

This filter supports the all above options as commands.

Apply high order Butterworth band-pass filter.

The filter accepts the following options:

Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.
Set filter order. Available values are from 4 to 20. Default value is 4.
Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.
Set input gain level. Allowed range is from 0 to 2. Default value is 1.

Commands

This filter supports the all above options as commands.

Apply high order Butterworth band-stop filter.

The filter accepts the following options:

Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.
Set filter order. Available values are from 4 to 20. Default value is 4.
Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.
Set input gain level. Allowed range is from 0 to 2. Default value is 1.

Commands

This filter supports the all above options as commands.

Adjust audio tempo.

The filter accepts exactly one parameter, the audio tempo. If not specified then the filter will assume nominal 1.0 tempo. Tempo must be in the [0.5, 100.0] range.

Note that tempo greater than 2 will skip some samples rather than blend them in. If for any reason this is a concern it is always possible to daisy-chain several instances of atempo to achieve the desired product tempo.

Examples

  • Slow down audio to 80% tempo:
    atempo=0.8
    
  • To speed up audio to 300% tempo:
    atempo=3
    
  • To speed up audio to 300% tempo by daisy-chaining two atempo instances:
    atempo=sqrt(3),atempo=sqrt(3)
    

Commands

This filter supports the following commands:

Change filter tempo scale factor. Syntax for the command is : "tempo"

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 the Time duration section in the ffmpeg-utils(1) manual.

Note that the first two sets of the start/end options and the duration option look at the frame timestamp, while the _sample options simply count the samples that pass through the filter. So start/end_pts and start/end_sample will give different results when the timestamps are wrong, inexact or do not start at zero. Also note that this filter does not modify the timestamps. If you wish to have the output timestamps start at zero, insert the asetpts filter after the atrim filter.

If multiple start or end options are set, this filter tries to be greedy and keep all samples that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple atrim filters.

The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time.

Examples:

  • Drop everything except the second minute of input:
    ffmpeg -i INPUT -af atrim=60:120
    
  • Keep only the first 1000 samples:
    ffmpeg -i INPUT -af atrim=end_sample=1000
    

Calculate normalized cross-correlation between two input audio streams.

Resulted samples are always between -1 and 1 inclusive. If result is 1 it means two input samples are highly correlated in that selected segment. Result 0 means they are not correlated at all. If result is -1 it means two input samples are out of phase, which means they cancel each other.

The filter accepts the following options:

Set size of segment over which cross-correlation is calculated. Default is 256. Allowed range is from 2 to 131072.
Set algorithm for cross-correlation. Can be "slow" or "fast". Default is "slow". Fast algorithm assumes mean values over any given segment are always zero and thus need much less calculations to make. This is generally not true, but is valid for typical audio streams.

Examples

Calculate correlation between channels in stereo audio stream:
ffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav

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.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

Commands

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"
Change bandpass mix. Syntax for the command is : "mix"

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