FFMPEG-FORMATS(1) FFMPEG-FORMATS(1)

ffmpeg-formats - FFmpeg formats

This document describes the supported formats (muxers and demuxers) provided by the libavformat library.

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 also set. In case the PTS is set, the DTS value is set to NOPTS. This is ignored when the "nofillin" flag 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.
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 if it requires an additional probing for PTS at end of file. At present, applicable for MPEG-PS and MPEG-TS.
Set probing size, in bytes, for input duration estimation when it actually requires an additional probing for PTS at end of file (at present: MPEG-PS and MPEG-TS). It is aimed at users interested in better durations probing for itself, or indirectly because using the concat demuxer, for example. The typical use case is an MPEG-TS CBR with a high bitrate, high video buffering and ending cleaning with similar PTS for video and audio: in such a scenario, the large physical gap between the last video packet and the last audio packet makes it necessary to read many bytes in order to get the video stream duration. Another use case is where the default probing behaviour only reaches a single video frame which is not the last one of the stream due to frame reordering, so the duration is not accurate. Setting this option has a performance impact even for small files because the probing size is fixed. Default behaviour is a general purpose trade-off, largely adaptive, but the probing size will not be extended to get streams durations at all costs. Must be an integer not lesser than 1, or 0 for default behaviour.
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.

Raw Audio Data Transport Stream AAC demuxer.

This demuxer is used to demux an ADTS input containing a single AAC stream alongwith any ID3v1/2 or APE tags in it.

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. Default is enabled.
Maximum framerate in frames per second. Default of 0 imposes no limit.
Default framerate in frames per second when none is specified in the file (0 meaning as fast as possible). Default is 15.

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.

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. This directive is deprecated, use "file_packet_meta" instead.
"file_packet_meta 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.
"option key value"
Option to access, open and probe the file. Can be present multiple times.
"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.
"stream_meta key value"
Metadata for the stream. Can be present multiple times.
"stream_codec value"
Codec for the stream.
"stream_extradata hex_string"
Extradata for the string, encoded in hexadecimal.
"chapter id start end"
Add a chapter. id is an unique identifier, possibly small and consecutive.

Options

This demuxer accepts the following option:

If set to 1, reject unsafe file paths and directives. 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.

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.

Options

This demuxer accepts the following option:

16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).

DVD-Video demuxer, powered by libdvdnav and libdvdread.

Can directly ingest DVD titles, specifically sequential PGCs, into a conversion pipeline. Menu assets, such as background video or audio, can also be demuxed given the menu's coordinates (at best effort). Seeking is not supported at this time.

Block devices (DVD drives), ISO files, and directory structures are accepted. Activate with "-f dvdvideo" in front of one of these inputs.

This demuxer does NOT have decryption code of any kind. You are on your own working with encrypted DVDs, and should not expect support on the matter.

Underlying playback is handled by libdvdnav, and structure parsing by libdvdread. FFmpeg must be built with GPL library support available as well as the configure switches "--enable-libdvdnav" and "--enable-libdvdread".

You will need to provide either the desired "title number" or exact PGC/PG coordinates. Many open-source DVD players and tools can aid in providing this information. If not specified, the demuxer will default to title 1 which works for many discs. However, due to the flexibility of the format, it is recommended to check manually. There are many discs that are authored strangely or with invalid headers.

If the input is a real DVD drive, please note that there are some drives which may silently fail on reading bad sectors from the disc, returning random bits instead which is effectively corrupt data. This is especially prominent on aging or rotting discs. A second pass and integrity checks would be needed to detect the corruption. This is not an FFmpeg issue.

Background

DVD-Video is not a directly accessible, linear container format in the traditional sense. Instead, it allows for complex and programmatic playback of carefully muxed MPEG-PS streams that are stored in headerless VOB files. To the end-user, these streams are known simply as "titles", but the actual logical playback sequence is defined by one or more "PGCs", or Program Group Chains, within the title. The PGC is in turn comprised of multiple "PGs", or Programs", which are the actual video segments (and for a typical video feature, sequentially ordered). The PGC structure, along with stream layout and metadata, are stored in IFO files that need to be parsed. PGCs can be thought of as playlists in easier terms.

An actual DVD player relies on user GUI interaction via menus and an internal VM to drive the direction of demuxing. Generally, the user would either navigate (via menus) or automatically be redirected to the PGC of their choice. During this process and the subsequent playback, the DVD player's internal VM also maintains a state and executes instructions that can create jumps to different sectors during playback. This is why libdvdnav is involved, as a linear read of the MPEG-PS blobs on the disc (VOBs) is not enough to produce the right sequence in many cases.

There are many other DVD structures (a long subject) that will not be discussed here. NAV packets, in particular, are handled by this demuxer to build accurate timing but not emitted as a stream. For a good high-level understanding, refer to: https://code.videolan.org/videolan/libdvdnav/-/blob/master/doc/dvd_structures

Options

This demuxer accepts the following options:

The title number to play. Must be set if pgc and pg are not set. Not applicable to menus. Default is 0 (auto), which currently only selects the first available title (title 1) and notifies the user about the implications.
The chapter, or PTT (part-of-title), number to start at. Not applicable to menus. Default is 1.
The chapter, or PTT (part-of-title), number to end at. Not applicable to menus. Default is 0, which is a special value to signal end at the last possible chapter.
The video angle number, referring to what is essentially an additional video stream that is composed from alternate frames interleaved in the VOBs. Not applicable to menus. Default is 1.
The region code to use for playback. Some discs may use this to default playback at a particular angle in different regions. This option will not affect the region code of a real DVD drive, if used as an input. Not applicable to menus. Default is 0, "world".
Demux menu assets instead of navigating a title. Requires exact coordinates of the menu (menu_lu, menu_vts, pgc, pg). Default is false.
The menu language to demux. In DVD, menus are grouped by language. Default is 0, the first language unit.
The VTS where the menu lives, or 0 if it is a VMG menu (root-level). Default is 0, VMG menu.
The entry PGC to start playback, in conjunction with pg. Alternative to setting title. Chapter markers are not supported at this time. Must be explicitly set for menus. Default is 0, automatically resolve from value of title.
The entry PG to start playback, in conjunction with pgc. Alternative to setting title. Chapter markers are not supported at this time. Default is 0, automatically resolve from value of title, or start from the beginning (PG 1) of the menu.
Enable this to have accurate chapter (PTT) markers and duration measurement, which requires a slow second pass read in order to index the chapter marker timestamps from NAV packets. This is non-ideal extra work for real optical drives. It is recommended and faster to use this option with a backup of the DVD structure stored on a hard drive. Not compatible with pgc and pg. Not applicable to menus. Default is 0, false.
Skip padding cells (i.e. cells shorter than 1 second) from the beginning. There exist many discs with filler segments at the beginning of the PGC, often with junk data intended for controlling a real DVD player's buffering speed and with no other material data value. Not applicable to menus. Default is 1, true.

Examples

  • Open title 3 from a given DVD structure:
    ffmpeg -f dvdvideo -title 3 -i <path to DVD> ...
    
  • Open chapters 3-6 from title 1 from a given DVD structure:
    ffmpeg -f dvdvideo -chapter_start 3 -chapter_end 6 -title 1 -i <path to DVD> ...
    
  • Open only chapter 5 from title 1 from a given DVD structure:
    ffmpeg -f dvdvideo -chapter_start 5 -chapter_end 5 -title 1 -i <path to DVD> ...
    
  • Demux menu with language 1 from VTS 1, PGC 1, starting at PG 1:
    ffmpeg -f dvdvideo -menu 1 -menu_lu 1 -menu_vts 1 -pgc 1 -pg 1 -i <path to DVD> ...
    

Electronic Arts Multimedia format demuxer.

This format is used by various Electronic Arts games.

Options

Normally the VP6 alpha channel (if exists) is returned as a secondary video stream, by setting this option you can make the demuxer return a single video stream which contains the alpha channel in addition to the ordinary video.

Interoperable Master Format demuxer.

This demuxer presents audio and video streams found in an IMF Composition, as specified in https://doi.org/10.5594/SMPTE.ST2067-2.2020.

ffmpeg [-assetmaps <path of ASSETMAP1>,<path of ASSETMAP2>,...] -i <path of CPL> ...

If "-assetmaps" is not specified, the demuxer looks for a file called ASSETMAP.xml in the same directory as the CPL.

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. KUX is a flv variant used on the Youku platform.

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).
prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
',' 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.
Set options for the demuxer of media segments using a list of key=value pairs separated by ":".
Maximum number of times to reload a segment on error, useful when segment skip on network error is not desired. Default value is 0.

Image file demuxer.

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

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

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

This demuxer accepts the following options:

Set the frame rate for the video stream. It defaults to 25.
If set to 1, loop over the input. Default value is 0.
Select the pattern type used to interpret the provided filename.

pattern_type accepts one of the following values.

Disable pattern matching, therefore the video will only contain the specified image. You should use this option if you do not want to create sequences from multiple images and your filenames may contain special pattern characters.
Select a sequence pattern type, used to specify a sequence of files indexed by sequential numbers.

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

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

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

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

ffmpeg -i img.jpeg img.png
Select a glob wildcard pattern type.

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

Select a mixed glob wildcard/sequence pattern.

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

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

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

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

Default value is glob_sequence.

Set the pixel format of the images to read. If not specified the pixel format is guessed from the first image file in the sequence.
Set the index of the file matched by the image file pattern to start to read from. Default value is 0.
Set the index interval range to check when looking for the first image file in the sequence, starting from start_number. Default value is 5.
If set to 1, will set frame timestamp to modification time of image file. Note that monotonity of timestamps is not provided: images go in the same order as without this option. Default value is 0. If set to 2, will set frame timestamp to the modification time of the image file in nanosecond precision.
Set the video size of the images to read. If not specified the video size is guessed from the first image file in the sequence.
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
For fragmented input, set fragment's starting timestamp to "baseMediaDecodeTime" from the "tfdt" box. Default is enabled, which will prefer to use the "tfdt" box to set DTS. Disable to use the "earliest_presentation_time" from the "sidx" box. In either case, the timestamp from the "mfra" box will be used if it's available and "use_mfra_for" is set to pts or dts.
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).
Very high sample deltas written in a trak's stts box may occasionally be intended but usually they are written in error or used to store a negative value for dts correction when treated as signed 32-bit integers. This option lets the user set an upper limit, beyond which the delta is clamped to 1. Values greater than the limit if negative when cast to int32 are used to adjust onward dts.

Unit is the track time scale. Range is 0 to UINT_MAX. Default is "UINT_MAX - 48000*10" which allows up to a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of "uint32" range.

Interleave packets from multiple tracks at demuxer level. For badly interleaved files, this prevents playback issues caused by large gaps between packets in different tracks, as MOV/MP4 do not have packet placement requirements. However, this can cause excessive seeking on very badly interleaved files, due to seeking between tracks, so disabling it may prevent I/O issues, at the expense of playback.

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.
Set maximum size, in bytes, of packet emitted by the demuxer. Payloads above this size are split across multiple packets. Range is 1 to INT_MAX/2. Default is 204800 bytes.

MJPEG encapsulated in multi-part MIME demuxer.

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

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

Raw video demuxer.

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

This demuxer accepts the following options:

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

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

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

RCWT (Raw Captions With Time) is a format native to ccextractor, a commonly used open source tool for processing 608/708 Closed Captions (CC) sources. For more information on the format, see .

This demuxer implements the specification as of March 2024, which has been stable and unchanged since April 2014.

Examples

  • Render CC to ASS using the built-in decoder:
    ffmpeg -i CC.rcwt.bin CC.ass
    

    Note that if your output appears to be empty, you may have to manually set the decoder's data_field option to pick the desired CC substream.

  • Convert an RCWT backup to Scenarist (SCC) format:
    ffmpeg -i CC.rcwt.bin -c:s copy CC.scc
    

    Note that the SCC format does not support all of the possible CC extensions that can be stored in RCWT (such as EIA-708).

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.

Sony Wave64 Audio demuxer.

This demuxer accepts the following options:

See the same option for the wav demuxer.

RIFF Wave Audio demuxer.

This demuxer accepts the following options:

Specify the maximum packet size in bytes for the demuxed packets. By default this is set to 0, which means that a sensible value is chosen based on the input format.

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.

This section covers raw muxers. They accept a single stream matching the designated codec. They do not store timestamps or metadata. The recognized extension is the same as the muxer name unless indicated otherwise.

It comprises the following muxers. The media type and the eventual extensions used to automatically selects the muxer from the output extensions are also shown.

Dolby Digital, also known as AC-3.
CRI Middleware ADX audio.

This muxer will write out the total sample count near the start of the first packet when the output is seekable and the count can be stored in 32 bits.

aptX (Audio Processing Technology for Bluetooth)
aptX HD (Audio Processing Technology for Bluetooth) audio
AVS2-P2 (Audio Video Standard - Second generation - Part 2) / IEEE 1857.4 video
AVS3-P2 (Audio Video Standard - Third generation - Part 2) / IEEE 1857.10 video
Chinese AVS (Audio Video Standard - First generation)
Codec 2 audio.

No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool "-f codec2raw".

Generic data muxer.

This muxer accepts a single stream with any codec of any type. The input stream has to be selected using the "-map" option with the ffmpeg CLI tool.

No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool "-f data".

Raw DFPWM1a (Dynamic Filter Pulse With Modulation) audio muxer.
BBC Dirac video.

The Dirac Pro codec is a subset and is standardized as SMPTE VC-2.

Avid DNxHD video.

It is standardized as SMPTE VC-3. Accepts DNxHR streams.

DTS Coherent Acoustics (DCA) audio
Dolby Digital Plus, also known as Enhanced AC-3
MPEG-5 Essential Video Coding (EVC) / EVC / MPEG-5 Part 1 EVC video
ITU-T G.722 audio
ITU-T G.723.1 audio
ITU-T G.726 big-endian ("left-justified") audio.

No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool "-f g726".

ITU-T G.726 little-endian ("right-justified") audio.

No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool "-f g726le".

Global System for Mobile Communications audio
ITU-T H.261 video
ITU-T H.263 / H.263-1996, H.263+ / H.263-1998 / H.263 version 2 video
ITU-T H.264 / MPEG-4 Part 10 AVC video. Bitstream shall be converted to Annex B syntax if it's in length-prefixed mode.
ITU-T H.265 / MPEG-H Part 2 HEVC video. Bitstream shall be converted to Annex B syntax if it's in length-prefixed mode.
MPEG-4 Part 2 video
Motion JPEG video
Meridian Lossless Packing, also known as Packed PCM
MPEG-1 Audio Layer II audio
MPEG-1 Part 2 video.
ITU-T H.262 / MPEG-2 Part 2 video
AV1 low overhead Open Bitstream Units muxer.

Temporal delimiter OBUs will be inserted in all temporal units of the stream.

rawvideo video (yuv, rgb)
Raw uncompressed video.
Bluetooth SIG low-complexity subband codec audio
Dolby TrueHD audio
SMPTE 421M / VC-1 video

Examples

Store raw video frames with the rawvideo muxer using ffmpeg:
ffmpeg -f lavfi -i testsrc -t 10 -s hd1080p testsrc.yuv

Since the rawvideo muxer do not store the information related to size and format, this information must be provided when demuxing the file:

ffplay -video_size 1920x1080 -pixel_format rgb24 -f rawvideo testsrc.rgb

This section covers raw PCM (Pulse-Code Modulation) audio muxers.

They accept a single stream matching the designated codec. They do not store timestamps or metadata. The recognized extension is the same as the muxer name.

It comprises the following muxers. The optional additional extension used to automatically select the muxer from the output extension is also shown in parentheses.

PCM A-law
PCM 32-bit floating-point big-endian
PCM 32-bit floating-point little-endian
PCM 64-bit floating-point big-endian
PCM 64-bit floating-point little-endian
PCM mu-law
PCM signed 16-bit big-endian
PCM signed 16-bit little-endian
PCM signed 24-bit big-endian
PCM signed 24-bit little-endian
PCM signed 32-bit big-endian
PCM signed 32-bit little-endian
PCM signed 8-bit
PCM unsigned 16-bit big-endian
PCM unsigned 16-bit little-endian
PCM unsigned 24-bit big-endian
PCM unsigned 24-bit little-endian
PCM unsigned 32-bit big-endian
PCM unsigned 32-bit little-endian
PCM unsigned 8-bit
PCM Archimedes VIDC

This section covers formats belonging to the MPEG-1 and MPEG-2 Systems family.

The MPEG-1 Systems format (also known as ISO/IEEC 11172-1 or MPEG-1 program stream) has been adopted for the format of media track stored in VCD (Video Compact Disc).

The MPEG-2 Systems standard (also known as ISO/IEEC 13818-1) covers two containers formats, one known as transport stream and one known as program stream; only the latter is covered here.

The MPEG-2 program stream format (also known as VOB due to the corresponding file extension) is an extension of MPEG-1 program stream: in addition to support different codecs for the audio and video streams, it also stores subtitles and navigation metadata. MPEG-2 program stream has been adopted for storing media streams in SVCD and DVD storage devices.

This section comprises the following muxers.

MPEG-1 Systems / MPEG-1 program stream muxer.
MPEG-1 Systems / MPEG-1 program stream (VCD) muxer.

This muxer can be used to generate tracks in the format accepted by the VCD (Video Compact Disc) storage devices.

It is the same as the mpeg muxer with a few differences.

MPEG-2 program stream (VOB) muxer.
MPEG-2 program stream (DVD VOB) muxer.

This muxer can be used to generate tracks in the format accepted by the DVD (Digital Versatile Disc) storage devices.

This is the same as the vob muxer with a few differences.

MPEG-2 program stream (SVCD VOB) muxer.

This muxer can be used to generate tracks in the format accepted by the SVCD (Super Video Compact Disc) storage devices.

This is the same as the vob muxer with a few differences.

Options

Set user-defined mux rate expressed as a number of bits/s. If not specied the automatically computed mux rate is employed. Default value is 0.
Set initial demux-decode delay in microseconds. Default value is 500000.

This section covers formats belonging to the QuickTime / MOV family, including the MPEG-4 Part 14 format and ISO base media file format (ISOBMFF). These formats share a common structure based on the ISO base media file format (ISOBMFF).

The MOV format was originally developed for use with Apple QuickTime. It was later used as the basis for the MPEG-4 Part 1 (later Part 14) format, also known as ISO/IEC 14496-1. That format was then generalized into ISOBMFF, also named MPEG-4 Part 12 format, ISO/IEC 14496-12, or ISO/IEC 15444-12.

It comprises the following muxers.

3gp
Third Generation Partnership Project (3GPP) format for 3G UMTS multimedia services
3g2
Third Generation Partnership Project 2 (3GP2 or 3GPP2) format for 3G CDMA2000 multimedia services, similar to 3gp with extensions and limitations
Adobe Flash Video format
MPEG-4 audio file format, as MOV/MP4 but limited to contain only audio streams, typically played with the Apple ipod device
Microsoft IIS (Internet Information Services) Smooth Streaming Audio/Video (ISMV or ISMA) format. This is based on MPEG-4 Part 14 format with a few incompatible variants, used to stream media files for the Microsoft IIS server.
QuickTime player format identified by the ".mov" extension
MP4 or MPEG-4 Part 14 format
PlayStation Portable MP4/MPEG-4 Part 14 format variant. This is based on MPEG-4 Part 14 format with a few incompatible variants, used to play files on PlayStation devices.

Fragmentation

The mov, mp4, and ismv muxers support fragmentation. Normally, a MOV/MP4 file has all the metadata about all packets stored in one location.

This data is usually written at the end of the file, but it can be moved to the start for better playback by adding "+faststart" to the "-movflags", or using the qt-faststart tool).

A fragmented file consists of a number of fragments, where packets and metadata about these packets are stored together. Writing a fragmented file has the advantage that the file is decodable even if the writing is interrupted (while a normal MOV/MP4 is undecodable if it is not properly finished), and it requires less memory when writing very long files (since writing normal MOV/MP4 files stores info about every single packet in memory until the file is closed). The downside is that it is less compatible with other applications.

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

If more than one condition is specified, fragments are cut when one of the specified conditions is fulfilled. The exception to this is the option min_frag_duration, which has to be fulfilled for any of the other conditions to apply.

Options

Override major brand.
Enable to skip writing the name inside a "hdlr" box. Default is "false".
set the media encryption key in hexadecimal format
set the media encryption key identifier in hexadecimal format
configure the encryption scheme, allowed values are none, and cenc-aes-ctr
Create fragments that are duration microseconds long.
Interleave samples within fragments (max number of consecutive samples, lower is tighter interleaving, but with more overhead. It is set to 0 by default.
create fragments that contain up to size bytes of payload data
specify iods number for the audio profile atom (from -1 to 255), default is -1
specify iods number for the video profile atom (from -1 to 255), default is -1
specify number of lookahead entries for ISM files (from 0 to 255), default is 0
do not create fragments that are shorter than duration microseconds long
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.
specify gamma value for gama atom (as a decimal number from 0 to 10), default is 0.0, must be set together with "+ movflags"
Set various muxing switches. The following flags can be used:
write CMAF (Common Media Application Format) compatible fragmented MP4 output
dash
write DASH (Dynamic Adaptive Streaming over HTTP) compatible fragmented MP4 output
Similarly to the omit_tfhd_offset flag, 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).
delay writing the initial moov until the first fragment is cut, or until the first fragment flush
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.
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.
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.)
signal that the next fragment is discontinuous from earlier ones
fragment at every frame
start a new fragment at each video keyframe
write a global sidx index at the start of the file
create a live smooth streaming feed (for pushing to a publishing point)
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.

Do not write any absolute base_data_offset in tfhd atoms. This avoids tying fragments to absolute byte positions in the file/streams.
If writing colr atom prioritise usage of ICC profile if it exists in stream packet side data.
add RTP hinting tracks to the output file
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.
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 the global_sidx flag is enabled, this option is ignored.
skip writing the mfra/tfra/mfro trailer for fragmented files
use mdta atom for metadata
write colr atom even if the color info is unspecified. This flag is experimental, may be renamed or changed, do not use from scripts.
write deprecated gama atom
For recoverability - write the output file as a fragmented file. This allows the intermediate file to be read while being written (in particular, if the writing process is aborted uncleanly). When writing is finished, the file is converted to a regular, non-fragmented file, which is more compatible and allows easier and quicker seeking.

If writing is aborted, the intermediate file can manually be remuxed to get a regular, non-fragmented file of what had been written into the unfinished file.

Set the timescale written in the movie header box ("mvhd"). Range is 1 to INT_MAX. Default is 1000.
Add RTP hinting tracks to the output file.

The following flags can be used:

use mode 0 for H.264 in RTP
use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC
use RFC 2190 packetization instead of RFC 4629 for H.263
send RTCP BYE packets when finishing
do not send RTCP sender reports
skip writing iods atom (default value is "true")
use edit list (default value is "auto")
use stream ids as track ids (default value is "false")
Set the timescale used for video tracks. Range is 0 to INT_MAX. If set to 0, the timescale is automatically set based on the native stream time base. Default is 0.
Force or disable writing bitrate box inside stsd box of a track. The box contains decoding buffer size (in bytes), maximum bitrate and average bitrate for the track. The box will be skipped if none of these values can be computed. Default is -1 or "auto", which will write the box only in MP4 mode.
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.
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).

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.

Examples

Push Smooth Streaming content in real time to a publishing point on IIS with the ismv muxer using ffmpeg:
ffmpeg -re <<normal input/transcoding options>> -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)

A64 Commodore 64 video muxer.

This muxer accepts a single "a64_multi" or "a64_multi5" codec video stream.

Raw AC-4 audio muxer.

This muxer accepts a single "ac4" audio stream.

Options

when enabled, write a CRC checksum for each packet to the output, default is "false"

Audio Data Transport Stream muxer.

It accepts a single AAC stream.

Options

Enable to write ID3v2.4 tags at the start of the stream. Default is disabled.
Enable to write APE tags at the end of the stream. Default is disabled.
Enable to set MPEG version bit in the ADTS frame header to 1 which indicates MPEG-2. Default is 0, which indicates MPEG-4.

MD STUDIO audio muxer.

This muxer accepts a single ATRAC1 audio stream with either one or two channels and a sample rate of 44100Hz.

As AEA supports storing the track title, this muxer will also write the title from stream's metadata to the container.

Audio Interchange File Format muxer.

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.

High Voltage Software's Lego Racers game audio muxer.

It accepts a single ADPCM_IMA_ALP stream with no more than 2 channels and a sample rate not greater than 44100 Hz.

Extensions: "tun", "pcm"

Options

Set file type.

type accepts the following values:

Set file type as music. Must have a sample rate of 22050 Hz.
Set file type as sfx.
Set file type as per output file extension. ".pcm" results in type "pcm" else type "tun" is set. (default)

3GPP AMR (Adaptive Multi-Rate) audio muxer.

It accepts a single audio stream containing an AMR NB stream.

AMV (Actions Media Video) format muxer.

Ubisoft Rayman 2 APM audio muxer.

It accepts a single ADPCM IMA APM audio stream.

Animated Portable Network Graphics muxer.

It accepts a single APNG video stream.

Options

Force a delay expressed in seconds after the last frame of each repetition. Default value is 0.0.
specify how many times to play the content, 0 causes an infinte loop, with 1 there is no loop

Examples

Use ffmpeg to generate an APNG output with 2 repetitions, and with a delay of half a second after the first repetition:
ffmpeg -i INPUT -final_delay 0.5 -plays 2 out.apng

Argonaut Games ASF audio muxer.

It accepts a single ADPCM audio stream.

Options

override file major version, specified as an integer, default value is 2
override file minor version, specified as an integer, default value is 1
Embed file name into file, if not specified use the output file name. The name is truncated to 8 characters.

Argonaut Games CVG audio muxer.

It accepts a single one-channel ADPCM 22050Hz audio stream.

The loop and reverb options set the corresponding flags in the header which can be later retrieved to process the audio stream accordingly.

Options

skip sample rate check (default is "false")
set loop flag (default is "false")
set reverb flag (default is "true")

Advanced / Active Systems (or Streaming) Format audio muxer.

The asf_stream variant should be selected for streaming.

Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this muxer too.

Options

Set the muxer packet size as a number of bytes. 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 "64Ki".

ASS/SSA (SubStation Alpha) subtitles muxer.

It accepts a single ASS subtitles stream.

Options

Write dialogue events immediately, even if they are out-of-order, default is "false", otherwise they are cached until the expected time event is found.

AST (Audio Stream) muxer.

This format is used to play audio on some Nintendo Wii games.

It accepts a single audio stream.

The loopstart and loopend options can be used to define a section of the file to loop for players honoring such options.

Options

Specify loop start position expressesd in milliseconds, from -1 to "INT_MAX", in case -1 is set then no loop is specified (default -1) and the loopend value is ignored.
Specify loop end position expressed in milliseconds, from 0 to "INT_MAX", default is 0, in case 0 is set it assumes the total stream duration.

SUN AU audio muxer.

It accepts a single audio stream.

Audio Video Interleaved muxer.

AVI is a proprietary format developed by Microsoft, and later formally specified through the Open DML specification.

Because of differences in players implementations, it might be required to set some options to make sure that the generated output can be correctly played by the target player.

Options

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.
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.

Default value is 0.

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).

AV1 (Alliance for Open Media Video codec 1) image format muxer.

This muxers stores images encoded using the AV1 codec.

It accepts one or two video streams. In case two video streams are provided, the second one shall contain a single plane storing the alpha mask.

In case more than one image is provided, the generated output is considered an animated AVIF and the number of loops can be specified with the loop option.

This is based on the specification by Alliance for Open Media at url https://aomediacodec.github.io/av1-avif.

Options

number of times to loop an animated AVIF, 0 specify an infinite loop, default is 0
Set the timescale written in the movie header box ("mvhd"). Range is 1 to INT_MAX. Default is 1000.

ShockWave Flash (SWF) / ActionScript Virtual Machine 2 (AVM2) format muxer.

It accepts one audio stream, one video stream, or both.

G.729 (.bit) file format muxer.

It accepts a single G.729 audio stream.

Apple CAF (Core Audio Format) muxer.

It accepts a single audio stream.

Codec2 audio audio muxer.

It accepts a single codec2 audio stream.

Chromaprint fingerprinter muxers.

To enable compilation of this filter you need to configure FFmpeg with "--enable-chromaprint".

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

Select 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:
Base64 compressed fingerprint (default)
Binary compressed fingerprint
Binary raw fingerprint
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.

CRC (Cyclic Redundancy Check) muxer.

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

  • Use ffmpeg to compute the CRC of the input, and store it in the file out.crc:
    ffmpeg -i INPUT -f crc out.crc
    
  • Use ffmpeg to 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 -
    

Dynamic Adaptive Streaming over HTTP (DASH) muxer.

This muxer creates segments and manifest files according to the MPEG-DASH standard ISO/IEC 23009-1:2014 and following standard updates.

For more information see:

This muxer creates an MPD (Media Presentation Description) manifest file and segment files for each stream. Segment files are placed in the same directory of the MPD manifest file.

The segment filename might contain pre-defined identifiers used in the manifest "SegmentTemplate" section 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.

Options

Assign streams to adaptation sets, specified in the MPD manifest "AdaptationSets" section.

An adaptation set contains a set of one or more streams accessed as a single subset, e.g. corresponding streams encoded at different size selectable by the user depending on the available bandwidth, or to different audio streams with a different language.

Each adaptation set is specified with the syntax:

id=<index>,streams=<streams>

where index must be a numerical index, and streams is a sequence of ","-separated stream indices. Multiple adaptation sets can be specified, separated by spaces.

To map all video (or audio) streams to an adaptation set, "v" (or "a") can be used as stream identifier instead of IDs.

When no assignment is defined, this defaults to an adaptation set for each stream.

The following optional fields can also be specified:

Define the descriptor as defined by ISO/IEC 23009-1:2014/Amd.2:2015.

For example:

<SupplementalProperty schemeIdUri=\"urn:mpeg:dash:srd:2014\" value=\"0,0,0,1,1,2,2\"/>

The descriptor string should be a self-closing XML tag.

Override the global fragment duration specified with the frag_duration option.
Override the global fragment type specified with the frag_type option.
Override the global segment duration specified with the seg_duration option.
Mark an adaptation set as containing streams meant to be used for Trick Mode for the referenced adaptation set.

A few examples of possible values for the adaptation_sets option follow:

id=0,seg_duration=2,frag_duration=1,frag_type=duration,streams=v id=1,seg_duration=2,frag_type=none,streams=a

id=0,seg_duration=2,frag_type=none,streams=0 id=1,seg_duration=10,frag_type=none,trick_id=0,streams=1
Set DASH segment files type.

Possible values:

The dash segment files format will be selected based on the stream codec. This is the default mode.
the dash segment files will be in ISOBMFF/MP4 format
the dash segment files will be in WebM format
Set the maximum number of segments kept outside of the manifest before removing from disk.
Set container format (mp4/webm) options using a ":"-separated list of key=value parameters. Values containing ":" special characters must be escaped.
Set the length in seconds of fragments within segments, fractional value can also be set.
Set the type of interval for fragmentation.

Possible values:

set one fragment per segment
fragment at every frame
fragment at specific time intervals
fragment at keyframes and following P-Frame reordering (Video only, experimental)
Write global "SIDX" atom. Applicable only for single file, mp4 output, non-streaming mode.
HLS master playlist name. Default is master.m3u8.
Generate HLS playlist files. The master playlist is generated with filename specified by the hls_master_name option. One media playlist file is generated for each stream with filenames media_0.m3u8, media_1.m3u8, etc.
Specify a list of ":"-separated key=value options to pass to the underlying HTTP protocol. Applicable only for HTTP output.
Use persistent HTTP connections. Applicable only for HTTP output.
Override User-Agent field in HTTP header. Applicable only for HTTP output.
Ignore IO errors during open and write. Useful for long-duration runs with network output. This is disabled by default.
Enable or disable segment index correction logic. Applicable only when use_template is enabled and use_timeline is disabled. This is disabled by default.

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.

DASH-templated name to use for the initialization segment. Default is "init-stream$RepresentationID$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format.
Enable Low-latency Dash by constraining the presence and values of some elements. This is disabled by default.
Enable Low-latency HLS (LHLS). Add "#EXT-X-PREFETCH" tag with current segment's URI. hls.js player folks are trying to standardize an open LHLS spec. The draft spec is available at https://github.com/video-dev/hlsjs-rfcs/blob/lhls-spec/proposals/0001-lhls.md.

This option tries to comply with the above open spec. It enables streaming and hls_playlist options automatically. This is an experimental feature.

Note: This is not Apple's version LHLS. See https://datatracker.ietf.org/doc/html/draft-pantos-hls-rfc8216bis

Publish master playlist repeatedly every after specified number of segment intervals.
Set the maximum playback rate indicated as appropriate for the purposes of automatically adjusting playback latency and buffer occupancy during normal playback by clients.
DASH-templated name to use 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.
Use the given HTTP method to create output files. Generally set to "PUT" or "POST".
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 one or more MPD manifest profiles.

Possible values:

dash
MPEG-DASH ISO Base media file format live profile
DVB-DASH profile

Default value is "dash".

Enable or disable removal of all segments when finished. This is disabled by default.
Set the segment length in seconds (fractional value can be set). The value is treated as average segment duration when the use_template option is enabled and the use_timeline option is disabled and as minimum segment duration for all the other use cases.

Default value is 5.

Enable or disable storing all segments in one file, accessed using byte ranges. This is disabled by default.

The name of the single file can be specified with the single_file_name option, if not specified assume the basename of the manifest file with the output format extension.

DASH-templated name to use for the manifest "baseURL" element. Imply that the single_file option is set to true. In the template, "$ext$" is replaced with the file name extension specific for the segment format.
Enable or disable chunk streaming mode of output. In chunk streaming mode, each frame will be a "moof" fragment which forms a chunk. This is disabled by default.
Set an intended target latency in seconds for serving (fractional value can be set). Applicable only when the streaming and write_prft options are enabled. This is an informative fields clients can use to measure the latency of the service.
Set timeout for socket I/O operations expressed in seconds (fractional value can be set). Applicable only for HTTP output.
Set the MPD update period, for dynamic content. The unit is second. If set to 0, the period is automatically computed.

Default value is 0.

Enable or disable use of "SegmentTemplate" instead of "SegmentList" in the manifest. This is enabled by default.
Enable or disable use of "SegmentTimeline" within the "SegmentTemplate" manifest section. This is enabled by default.
URL of the page that will return the UTC timestamp in ISO format, for example "https://time.akamai.com/?iso"
Set the maximum number of segments kept in the manifest, discard the oldest one. This is useful for live streaming.

If the value is 0, all segments are kept in the manifest. Default value is 0.

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 is set to auto by default, in which case the muxer will attempt to enable it only in modes that require it.

Example

Generate a DASH output reading from an input source in realtime using ffmpeg.

Two multimedia streams are generated from the input file, both containing a video stream encoded through libx264, and an audio stream encoded with libfdk_aac. The first multimedia stream contains video with a bitrate of 800k and audio at the default rate, the second with video scaled to 320x170 pixels at 300k and audio resampled at 22005 Hz.

The window_size option keeps only the latest 5 segments with the default duration of 5 seconds.

ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264 \
-b:v:0 800k -profile:v:0 main \
-b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline -ar:a:1 22050 \
-bf 1 -keyint_min 120 -g 120 -sc_threshold 0 -b_strategy 0 \
-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

D-Cinema audio muxer.

It accepts a single 6-channels audio stream resampled at 96000 Hz encoded with the pcm_24daud codec.

Example

Use ffmpeg to mux input audio to a 5.1 channel layout resampled at 96000Hz:

ffmpeg -i INPUT -af aresample=96000,pan=5.1 slow.302

For ffmpeg versions before 7.0 you might have to use the asetnsamples filter to limit the muxed packet size, because this format does not support muxing packets larger than 65535 bytes (3640 samples). For newer ffmpeg versions audio is automatically packetized to 36000 byte (2000 sample) packets.

DV (Digital Video) muxer.

It accepts exactly one dvvideo video stream and at most two pcm_s16 audio streams. More constraints are defined by the property of the video, which must correspond to a DV video supported profile, and on the framerate.

Example

Use ffmpeg to convert the input:

ffmpeg -i INPUT -s:v 720x480 -pix_fmt yuv411p -r 29.97 -ac 2 -ar 48000 -y out.dv

FFmpeg metadata muxer.

This muxer writes the streams metadata in the ffmetadata format.

See the Metadata chapter for information about the format.

Example

Use ffmpeg to extract metadata from an input file to a metadata.ffmeta file in ffmetadata format:

ffmpeg -i INPUT -f ffmetadata metadata.ffmeta

FIFO (First-In First-Out) muxer.

The fifo pseudo-muxer allows the separation of encoding and muxing by using a 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.

The target muxer is either selected from the output name or specified through the fifo_format option.

The behavior of the fifo muxer if the queue fills up or if the output fails (e.g. if a packet cannot be written to the output) 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 the FIFO queue fills up.

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

Options

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 "false".
If set to "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 "false", so in such cases the encoder will be blocked until the muxer processes some of the packets and none of them is lost.
Specify the format name. Useful if it cannot be guessed from the output name suffix.
Specify format options for the underlying muxer. Muxer options can be specified as a list of key=value pairs separated by ':'.
Set maximum number of successive unsuccessful recovery attempts after which the output fails permanently. By default this option is set to 0 (unlimited).
Specify size of the queue as a number of packets. Default value is 60.
If set to "true", recovery will be attempted regardless of type of the error causing the failure. By default this option is set to "false" and in case of certain (usually permanent) errors the recovery is not attempted even when the attempt_recovery option is set to "true".
If set to "false", the real time is used when waiting for the recovery attempt (i.e. the recovery will be attempted after the time specified by the recovery_wait_time option).

If set to "true", the time of the processed stream is taken into account instead (i.e. the recovery will be attempted after discarding the packets corresponding to the recovery_wait_time option).

By default this option is set to "false".

Specify waiting time in seconds before the next recovery attempt after previous unsuccessful recovery attempt. Default value is 5.
Specify whether to wait for the keyframe after recovering from queue overflow or failure. This option is set to "false" by default.
Buffer the specified amount of packets and delay writing the output. Note that the value of the queue_size option must be big enough to store the packets for timeshift. At the end of the input the fifo buffer is flushed at realtime speed.

Example

Use ffmpeg to stream to an 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 \
  -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 \
  -map 0:v -map 0:a rtmp://example.com/live/stream_name

Sega film (.cpk) muxer.

This format was used as internal format for several Sega games.

For more information regarding the Sega film file format, visit http://wiki.multimedia.cx/index.php?title=Sega_FILM.

It accepts at maximum one cinepak or raw video stream, and at maximum one audio stream.

Adobe Filmstrip muxer.

This format is used by several Adobe tools to store a generated filmstrip export. It accepts a single raw video stream.

Flexible Image Transport System (FITS) muxer.

This image format is used to store astronomical data.

For more information regarding the format, visit https://fits.gsfc.nasa.gov.

Raw FLAC audio muxer.

This muxer accepts exactly one FLAC audio stream. Additionally, it is possible to add images with disposition attached_pic.

Options

write the file header if set to "true", default is "true"

Example

Use ffmpeg to store the audio stream from an input file, together with several pictures used with attached_pic disposition:

ffmpeg -i INPUT -i pic1.png -i pic2.jpg -map 0:a -map 1 -map 2 -disposition:v attached_pic OUTPUT

Adobe Flash Video Format muxer.

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.

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.

Note that the GIF format has a very large time base: the delay between two frames can therefore not be smaller than one centi second.

Options

Set the number of times to loop the output. Use -1 for no loop, 0 for looping indefinitely (default).
Force the delay (expressed in centiseconds) after the last frame. Each frame ends with a delay until the next frame. The default is -1, which is a special value to tell the muxer to re-use the previous delay. In case of a loop, you might want to customize this value to mark a pause for instance.

Example

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"

General eXchange Format (GXF) muxer.

GXF was developed by Grass Valley Group, then standardized by SMPTE as SMPTE 360M and was extended in SMPTE RDD 14-2007 to include high-definition video resolutions.

It accepts at most one video stream with codec mjpeg, or mpeg1video, or mpeg2video, or dvvideo with resolution 512x480 or 608x576, and several audio streams with rate 48000Hz and codec pcm16_le.

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.

HTTP Dynamic Streaming (HDS) muxer.

HTTP dynamic streaming, or HDS, is an adaptive bitrate streaming method developed by Adobe. HDS delivers MP4 video content over HTTP connections. HDS can be used for on-demand streaming or live streaming.

This muxer creates an .f4m (Adobe Flash Media Manifest File) manifest, an .abst (Adobe Bootstrap File) for each stream, and segment files in a directory specified as the output.

These needs to be accessed by an HDS player throuhg HTTPS for it to be able to perform playback on the generated stream.

Options

number of fragments kept outside of the manifest before removing from disk
minimum fragment duration (in microseconds), default value is 1 second (10000000)
remove all fragments when finished when set to "true"
number of fragments kept in the manifest, if set to a value different from 0. By default all segments are kept in the output directory.

Example

Use ffmpeg to generate HDS files to the output.hds directory in real-time rate:

ffmpeg -re -i INPUT -f hds -b:v 200k output.hds

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

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.
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 start numbers according to the start_number option value.
Set the start number as the seconds since epoch (1970-01-01 00:00:00).
Set the start number as the microseconds since epoch (1970-01-01 00:00:00).
Set the start number 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 the hls_flags option is set with single_file, filename is used as a string format with the segment number appended.

For example:

ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8

will produce the playlist, out.m3u8, and segment files: file000.ts, file001.ts, file002.ts, etc.

filename may contain a full path or relative path specification, but only the file name part without any path 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", and this string will be expanded to the position of variant stream index in the generated segment file names.

For example:

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

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.

For example:

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

will produce the playlists segment file sets: vs0/file_000.ts, vs0/file_001.ts, vs0/file_002.ts, etc. and vs1/file_000.ts, vs1/file_001.ts, vs1/file_002.ts, etc.

Use strftime() on filename to expand the segment filename with localtime. The segment number is also available in this mode, but to use it, you need to set second_level_segment_index in the hls_flag and %%d will be the specifier.

For example:

ffmpeg -i in.nut -strftime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8

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.

For example:

ffmpeg -i in.nut -strftime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8

will produce the playlist, out.m3u8, and segment files: file-20160215-0001.ts, file-20160215-0002.ts, etc.

Used together with strftime, it will create all subdirectories which are present in the expanded values of option hls_segment_filename.

For example:

ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8

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.

For example:

ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8

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.

Set output format options using a :-separated list of key=value parameters. Values containing ":" special characters must be escaped.
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.
Specify a 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.
Specify the 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 for the fragment files header file, default filename is init.mp4.

When strftime is enabled, filename is expanded to the segment filename with localtime.

For example:

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

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. If there are no other features requiring higher HLS versions be used, then this will allow ffmpeg to output a HLS version 2 m3u8.
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" tag 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" tag 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 key frames. This improves behavior on some players when the time between key frames 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.
Make it possible to use segment indexes as %%d in the hls_segment_filename option expression besides date/time values when strftime option is on. To get fixed width numbers with trailing zeroes, %%0xd format is available where x is the required width.
Make it possible to use segment sizes (counted in bytes) as %%s in hls_segment_filename option 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.
Make it possible to use segment duration (calculated in microseconds) as %%t in hls_segment_filename option 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.

For example:

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

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 be written into a temporary file and renamed after they are complete, similarly as segments are handled. But playlists with "file" protocol and with hls_playlist_type type other than vod are always written into a temporary file regardless of this flag.

Master playlist files specified with 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.

If type is event, emit "#EXT-X-PLAYLIST-TYPE:EVENT" in the m3u8 header. This forces hls_list_size to 0; the playlist can only be appended to.

If type is vod, emit "#EXT-X-PLAYLIST-TYPE:VOD" in the m3u8 header. This forces hls_list_size to 0; the playlist must not change.

Use the given HTTP method to create the hls files.

For example:

ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8

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.
Specify a map string defining 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.

A few examples follow.

  • Create two hls variant streams. The first variant stream will contain video stream of bitrate 1000k and audio stream of bitrate 64k and the second variant stream will contain video stream of bitrate 256k and audio stream of bitrate 32k. Here, two media playlist with file names out_0.m3u8 and out_1.m3u8 will be created.
    ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -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
    
  • If you want something meaningful text instead of indexes in result names, you may specify names for each or some of the variants. The following example will create 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 -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
    
  • Create 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 \
      -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
    
  • Create 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: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
    
  • Create 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, the "#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 -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
    
  • Create 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, the "#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 a:1,agroup:aud_low v:0,agroup:aud_low" \
      -master_pl_name master.m3u8 \
      http://example.com/live/out_%v.m3u8
    
  • Create 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, the "#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 -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
    
  • Create a single variant stream. Add the "#EXT-X-MEDIA" tag with "TYPE=SUBTITLES" in the master playlist with webvtt subtitle group name 'subtitle'. Make sure the input file has one text subtitle stream at least.
    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
    
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.

For example:

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

will add two "#EXT-X-MEDIA" tags with "TYPE=CLOSED-CAPTIONS" in the master playlist for the INSTREAM-IDs 'CC1' and 'CC2'. Also, it will add "CLOSED-CAPTIONS" attribute with group name 'cc' for the two output variant streams.

If var_stream_map is not set, then the first available ccgroup in cc_stream_map is mapped to the output variant stream.

For example:

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 will add "#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 will add "CLOSED-CAPTIONS" attribute with group name 'cc' for the output variant stream.

Create HLS master playlist with the given name.

For example:

ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 http://example.com/live/out.m3u8

creates an HLS master playlist with name master.m3u8 which is published at http://example.com/live/.

Publish master play list repeatedly every after specified number of segment intervals.

For example:

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

creates an HLS master playlist with name master.m3u8 and keeps 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.

Immersive Audio Model and Formats (IAMF) muxer.

IAMF is used to provide immersive audio content for presentation on a wide range of devices in both streaming and offline applications. These applications include internet audio streaming, multicasting/broadcasting services, file download, gaming, communication, virtual and augmented reality, and others. In these applications, audio may be played back on a wide range of devices, e.g., headphones, mobile phones, tablets, TVs, sound bars, home theater systems, and big screens.

This format was promoted and desgined by Alliance for Open Media.

For more information about this format, see https://aomedia.org/iamf/.

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

Internet Low Bitrate Codec (iLBC) raw muxer.

It accepts a single ilbc audio stream.

Image file muxer.

The image2 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 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.

The image2pipe muxer accepts the same options as the image2 muxer, but ignores the pattern verification and expansion, as it is supposed to write to the command output rather than to an actual stored file.

Options

If set to 1, expand the filename with the packet PTS (presentation time stamp). 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.
Write output to a temporary file, which is renamed to target filename once writing is completed. Default is disabled.
Set protocol options as a :-separated list of key=value parameters. Values containing the ":" special character must be escaped.

Examples

  • 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.

    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"
    
  • 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
    
  • 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
    

Berkeley / IRCAM / CARL Sound Filesystem (BICSF) format muxer.

The Berkeley/IRCAM/CARL Sound Format, developed in the 1980s, is a result of the merging of several different earlier sound file formats and systems including the csound system developed by Dr Gareth Loy at the Computer Audio Research Lab (CARL) at UC San Diego, the IRCAM sound file system developed by Rob Gross and Dan Timis at the Institut de Recherche et Coordination Acoustique / Musique in Paris and the Berkeley Fast Filesystem.

It was developed initially as part of the Berkeley/IRCAM/CARL Sound Filesystem, a suite of programs designed to implement a filesystem for audio applications running under Berkeley UNIX. It was particularly popular in academic music research centres, and was used a number of times in the creation of early computer-generated compositions.

This muxer accepts a single audio stream containing PCM data.

On2 IVF muxer.

IVF was developed by On2 Technologies (formerly known as Duck Corporation), to store internally developed codecs.

This muxer accepts a single vp8, vp9, or av1 video stream.

JACOsub subtitle format muxer.

This muxer accepts a single jacosub subtitles stream.

For more information about the format, see http://unicorn.us.com/jacosub/jscripts.html.

Simon & Schuster Interactive VAG muxer.

This custom VAG container is used by some Simon & Schuster Interactive games such as "Real War", and "Real War: Rogue States".

This muxer accepts a single adpcm_ima_ssi audio stream.

Bluetooth SIG Low Complexity Communication Codec audio (LC3), or ETSI TS 103 634 Low Complexity Communication Codec plus (LC3plus).

This muxer accepts a single lc3 audio stream.

LRC lyrics file format muxer.

LRC (short for LyRiCs) is a computer file format that synchronizes song lyrics with an audio file, such as MP3, Vorbis, or MIDI.

This muxer accepts a single subrip or text subtitles stream.

Metadata

The following metadata tags are converted to the format corresponding metadata:

If encoder_version is not explicitly set, it is automatically set to the libavformat version.

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

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 size bytes 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.

If set, the muxer will write the index at the beginning of the file by shifting the main data if necessary. This can be combined with reserve_index_space in which case the data is only shifted if the initially reserved space turns out to be insufficient.

This option is ignored if the output is unseekable.

Store at most the provided amount of bytes in a cluster.

If not specified, the limit is set automatically to a sensible hardcoded fixed value.

Store at most the provided number of milliseconds in a cluster.

If not specified, the limit is set automatically to a sensible hardcoded fixed value.

dash bool
Create a WebM file conforming to WebM DASH specification. By default it is set to "false".
Track number for the DASH stream. By default it is set to 1.
Write files assuming it is a live stream. By default it is set to "false".
Allow raw VFW mode. By default it is set to "false".
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.
Write a CRC32 element inside every Level 1 element. By default it is set to "true". This option is ignored for WebM.
Control how the FlagDefault of the output tracks will be set. It influences which tracks players should play by default. The default mode is passthrough.
Every track with disposition default will have the FlagDefault set. Additionally, for each type of track (audio, video or subtitle), if no track with disposition default of this type exists, then the first track of this type will be marked as default (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.

MD5 testing format.

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

See also the hash and framemd5 muxers.

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
    
  • To print the MD5 hash to stdout:
    ffmpeg -i INPUT -f md5 -
    

MicroDVD subtitle format muxer.

This muxer accepts a single microdvd subtitles stream.

Synthetic music Mobile Application Format (SMAF) format muxer.

SMAF is a music data format specified by Yamaha for portable electronic devices, such as mobile phones and personal digital assistants.

This muxer accepts a single adpcm_yamaha audio stream.

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.
Emit NIT table.
Disable writing of random access indicator.
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.
Maximum time in seconds between NIT tables. Default is 0.5.
Set PAT, PMT, SDT and NIT 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.

RCWT (Raw Captions With Time) is a format native to ccextractor, a commonly used open source tool for processing 608/708 Closed Captions (CC) sources. It can be used to archive the original extracted CC bitstream and to produce a source file for later processing or conversion. The format allows for interoperability between ccextractor and FFmpeg, is simple to parse, and can be used to create a backup of the CC presentation.

This muxer implements the specification as of March 2024, which has been stable and unchanged since April 2014.

This muxer will have some nuances from the way that ccextractor muxes RCWT. No compatibility issues when processing the output with ccextractor have been observed as a result of this so far, but mileage may vary and outputs will not be a bit-exact match.

A free specification of RCWT can be found here: https://github.com/CCExtractor/ccextractor/blob/master/docs/BINARY_FILE_FORMAT.TXT

Examples

Extract Closed Captions to RCWT using lavfi:
ffmpeg -f lavfi -i "movie=INPUT.mkv[out+subcc]" -map 0:s:0 -c:s copy -f rcwt CC.rcwt.bin

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:

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.

Set minimum segment duration to time, the value must be a duration specification. This prevents the muxer ending segments at a duration below this value. Only effective with "segment_time". Default value is "0".
If set to "1" split at regular clock time intervals starting from 00:00 o'clock. The time value specified in segment_time is used for setting the length of the splitting interval.

For example with segment_time set to "900" this makes it possible to create files at 12:00 o'clock, 12:15, 12:30, etc.

Default value is "0".

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 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 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

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

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

ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)

The FFmpeg developers.

For details about the authorship, see the Git history of the project (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in the FFmpeg source directory, or browsing the online repository at https://git.ffmpeg.org/ffmpeg.

Maintainers for the specific components are listed in the file MAINTAINERS in the source code tree.