# FFMPEG-ALL - ffmpeg media converter - man - [phpMan] [_FFMPEG-ALL_(1)](https://www.chedong.com/phpMan.php/man/FFMPEG-ALL/1/markdown) [_FFMPEG-ALL_(1)](https://www.chedong.com/phpMan.php/man/FFMPEG-ALL/1/markdown) ## NAME ffmpeg - ffmpeg media converter ## SYNOPSIS ffmpeg [_global_options_] {[_input_file_options_] -i _input_url_} ... {[_output_file_options_] _output_url_} ... ## DESCRIPTION **ffmpeg **is a universal media converter. It can read a wide variety of inputs - including live grabbing/recording devices - filter, and transcode them into a plethora of output formats. **ffmpeg **reads from an arbitrary number of input "files" (which can be regular files, pipes, network streams, grabbing devices, etc.), specified by the "-i" option, and writes to an arbitrary number of output "files", which are specified by a plain output url. Anything found on the command line which cannot be interpreted as an option is considered to be an output url. Each input or output url can, in principle, contain any number of streams of different types (video/audio/subtitle/attachment/data). The allowed number and/or types of streams may be limited by the container format. Selecting which streams from which inputs will go into which output is either done automatically or with the "-map" option (see the Stream selection chapter). To refer to input files in options, you must use their indices (0-based). E.g. the first input file is 0, the second is 1, etc. Similarly, streams within a file are referred to by their indices. E.g. "2:3" refers to the fourth stream in the third input file. Also see the Stream specifiers chapter. As a general rule, options are applied to the next specified file. Therefore, order is important, and you can have the same option on the command line multiple times. Each occurrence is then applied to the next input or output file. Exceptions from this rule are the global options (e.g. verbosity level), which should be specified first. Do not mix input and output files -- first specify all input files, then all output files. Also do not mix options which belong to different files. All options apply ONLY to the next input or output file and are reset between files. Some simple examples follow. • Convert an input media file to a different format, by re-encoding media streams: ffmpeg -i input.avi output.mp4 • Set the video bitrate of the output file to 64 kbit/s: ffmpeg -i input.avi -b:v 64k -bufsize 64k output.mp4 • Force the frame rate of the output file to 24 fps: ffmpeg -i input.avi -r 24 output.mp4 • Force the frame rate of the input file (valid for raw formats only) to 1 fps and the frame rate of the output file to 24 fps: ffmpeg -r 1 -i input.m2v -r 24 output.mp4 The format option may be needed for raw input files. ## DETAILED DESCRIPTION The transcoding process in **ffmpeg **for each output can be described by the following diagram: _______ ______________ | | | | | input | demuxer | encoded data | decoder | file | ---------> | packets | -----+ |_______| |______________| | v _________ | | | decoded | | frames | |_________| ________ ______________ | | | | | | | output | <-------- | encoded data | <----+ | file | muxer | packets | encoder |________| |______________| **ffmpeg **calls the libavformat library (containing demuxers) to read input files and get packets containing encoded data from them. When there are multiple input files, **ffmpeg **tries to keep them synchronized by tracking lowest timestamp on any active input stream. Encoded packets are then passed to the decoder (unless streamcopy is selected for the stream, see further for a description). The decoder produces uncompressed frames (raw video/PCM audio/...) which can be processed further by filtering (see next section). After filtering, the frames are passed to the encoder, which encodes them and outputs encoded packets. Finally those are passed to the muxer, which writes the encoded packets to the output file. ### Filtering Before encoding, **ffmpeg **can process raw audio and video frames using filters from the libavfilter library. Several chained filters form a filter graph. **ffmpeg **distinguishes between two types of filtergraphs: simple and complex. _Simple_ _filtergraphs_ Simple filtergraphs are those that have exactly one input and output, both of the same type. In the above diagram they can be represented by simply inserting an additional step between decoding and encoding: _________ ______________ | | | | | decoded | | encoded data | | frames |\ _ | packets | |_________| \ /||______________| \ __________ / simple _\|| | / encoder filtergraph | filtered |/ | frames | |__________| Simple filtergraphs are configured with the per-stream **-filter **option (with **-vf **and **-af** aliases for video and audio respectively). A simple filtergraph for video can look for example like this: _______ _____________ _______ ________ | | | | | | | | | input | ---> | deinterlace | ---> | scale | ---> | output | |_______| |_____________| |_______| |________| Note that some filters change frame properties but not frame contents. E.g. the "fps" filter in the example above changes number of frames, but does not touch the frame contents. Another example is the "setpts" filter, which only sets timestamps and otherwise passes the frames unchanged. _Complex_ _filtergraphs_ Complex filtergraphs are those which cannot be described as simply a linear processing chain applied to one stream. This is the case, for example, when the graph has more than one input and/or output, or when output stream type is different from input. They can be represented with the following diagram: _________ | | | input 0 |\ __________ |_________| \ | | \ _________ /| output 0 | \ | | / |__________| _________ \| complex | / | | | |/ | input 1 |---->| filter |\ |_________| | | \ __________ /| graph | \ | | / | | \| output 1 | _________ / |_________| |__________| | | / | input 2 |/ |_________| Complex filtergraphs are configured with the **-filter_complex **option. Note that this option is global, since a complex filtergraph, by its nature, cannot be unambiguously associated with a single stream or file. The **-lavfi **option is equivalent to **-filter_complex**. A trivial example of a complex filtergraph is the "overlay" filter, which has two video inputs and one video output, containing one video overlaid on top of the other. Its audio counterpart is the "amix" filter. ### Stream copy Stream copy is a mode selected by supplying the "copy" parameter to the **-codec **option. It makes **ffmpeg **omit the decoding and encoding step for the specified stream, so it does only demuxing and muxing. It is useful for changing the container format or modifying container- level metadata. The diagram above will, in this case, simplify to this: _______ ______________ ________ | | | | | | | input | demuxer | encoded data | muxer | output | | file | ---------> | packets | -------> | file | |_______| |______________| |________| Since there is no decoding or encoding, it is very fast and there is no quality loss. However, it might not work in some cases because of many factors. Applying filters is obviously also impossible, since filters work on uncompressed data. ## STREAM SELECTION **ffmpeg **provides the "-map" option for manual control of stream selection in each output file. Users can skip "-map" and let ffmpeg perform automatic stream selection as described below. The "-vn / -an / -sn / -dn" options can be used to skip inclusion of video, audio, subtitle and data streams respectively, whether manually mapped or automatically selected, except for those streams which are outputs of complex filtergraphs. ### Description The sub-sections that follow describe the various rules that are involved in stream selection. The examples that follow next show how these rules are applied in practice. While every effort is made to accurately reflect the behavior of the program, FFmpeg is under continuous development and the code may have changed since the time of this writing. _Automatic_ _stream_ _selection_ In the absence of any map options for a particular output file, ffmpeg inspects the output format to check which type of streams can be included in it, viz. video, audio and/or subtitles. For each acceptable stream type, ffmpeg will pick one stream, when available, from among all the inputs. It will select that stream based upon the following criteria: • for video, it is the stream with the highest resolution, • for audio, it is the stream with the most channels, • for subtitles, it is the first subtitle stream found but there's a caveat. The output format's default subtitle encoder can be either text-based or image-based, and only a subtitle stream of the same type will be chosen. In the case where several streams of the same type rate equally, the stream with the lowest index is chosen. Data or attachment streams are not automatically selected and can only be included using "-map". _Manual_ _stream_ _selection_ When "-map" is used, only user-mapped streams are included in that output file, with one possible exception for filtergraph outputs described below. _Complex_ _filtergraphs_ If there are any complex filtergraph output streams with unlabeled pads, they will be added to the first output file. This will lead to a fatal error if the stream type is not supported by the output format. In the absence of the map option, the inclusion of these streams leads to the automatic stream selection of their types being skipped. If map options are present, these filtergraph streams are included in addition to the mapped streams. Complex filtergraph output streams with labeled pads must be mapped once and exactly once. _Stream_ _handling_ Stream handling is independent of stream selection, with an exception for subtitles described below. Stream handling is set via the "-codec" option addressed to streams within a specific _output_ file. In particular, codec options are applied by ffmpeg after the stream selection process and thus do not influence the latter. If no "-codec" option is specified for a stream type, ffmpeg will select the default encoder registered by the output file muxer. An exception exists for subtitles. If a subtitle encoder is specified for an output file, the first subtitle stream found of any type, text or image, will be included. ffmpeg does not validate if the specified encoder can convert the selected stream or if the converted stream is acceptable within the output format. This applies generally as well: when the user sets an encoder manually, the stream selection process cannot check if the encoded stream can be muxed into the output file. If it cannot, ffmpeg will abort and _all_ output files will fail to be processed. ### Examples The following examples illustrate the behavior, quirks and limitations of ffmpeg's stream selection methods. They assume the following three input files. input file 'A.avi' stream 0: video 640x360 stream 1: audio 2 channels input file 'B.mp4' stream 0: video 1920x1080 stream 1: audio 2 channels stream 2: subtitles (text) stream 3: audio 5.1 channels stream 4: subtitles (text) input file 'C.mkv' stream 0: video 1280x720 stream 1: audio 2 channels stream 2: subtitles (image) Example: automatic stream selection ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov There are three output files specified, and for the first two, no "-map" options are set, so ffmpeg will select streams for these two files automatically. _out1.mkv_ is a Matroska container file and accepts video, audio and subtitle streams, so ffmpeg will try to select one of each type.For video, it will select "stream 0" from _B.mp4_, which has the highest resolution among all the input video streams.For audio, it will select "stream 3" from _B.mp4_, since it has the greatest number of channels.For subtitles, it will select "stream 2" from _B.mp4_, which is the first subtitle stream from among _A.avi_ and _B.mp4_. _out2.wav_ accepts only audio streams, so only "stream 3" from _B.mp4_ is selected. For _out3.mov_, since a "-map" option is set, no automatic stream selection will occur. The "-map 1:a" option will select all audio streams from the second input _B.mp4_. No other streams will be included in this output file. For the first two outputs, all included streams will be transcoded. The encoders chosen will be the default ones registered by each output format, which may not match the codec of the selected input streams. For the third output, codec option for audio streams has been set to "copy", so no decoding- filtering-encoding operations will occur, or _can_ occur. Packets of selected streams shall be conveyed from the input file and muxed within the output file. Example: automatic subtitles selection ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv Although _out1.mkv_ is a Matroska container file which accepts subtitle streams, only a video and audio stream shall be selected. The subtitle stream of _C.mkv_ is image-based and the default subtitle encoder of the Matroska muxer is text-based, so a transcode operation for the subtitles is expected to fail and hence the stream isn't selected. However, in _out2.mkv_, a subtitle encoder is specified in the command and so, the subtitle stream is selected, in addition to the video stream. The presence of "-an" disables audio stream selection for _out2.mkv_. Example: unlabeled filtergraph outputs ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt A filtergraph is setup here using the "-filter_complex" option and consists of a single video filter. The "overlay" filter requires exactly two video inputs, but none are specified, so the first two available video streams are used, those of _A.avi_ and _C.mkv_. The output pad of the filter has no label and so is sent to the first output file _out1.mp4_. Due to this, automatic selection of the video stream is skipped, which would have selected the stream in _B.mp4_. The audio stream with most channels viz. "stream 3" in _B.mp4_, is chosen automatically. No subtitle stream is chosen however, since the MP4 format has no default subtitle encoder registered, and the user hasn't specified a subtitle encoder. The 2nd output file, _out2.srt_, only accepts text-based subtitle streams. So, even though the first subtitle stream available belongs to _C.mkv_, it is image-based and hence skipped. The selected stream, "stream 2" in _B.mp4_, is the first text-based subtitle stream. Example: labeled filtergraph outputs ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \ -map '[outv]' -an out1.mp4 \ out2.mkv \ -map '[outv]' -map 1:a:0 out3.mkv The above command will fail, as the output pad labelled "[outv]" has been mapped twice. None of the output files shall be processed. ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \ -an out1.mp4 \ out2.mkv \ -map 1:a:0 out3.mkv This command above will also fail as the hue filter output has a label, "[outv]", and hasn't been mapped anywhere. The command should be modified as follows, ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \ -map '[outv1]' -an out1.mp4 \ out2.mkv \ -map '[outv2]' -map 1:a:0 out3.mkv The video stream from _B.mp4_ is sent to the hue filter, whose output is cloned once using the split filter, and both outputs labelled. Then a copy each is mapped to the first and third output files. The overlay filter, requiring two video inputs, uses the first two unused video streams. Those are the streams from _A.avi_ and _C.mkv_. The overlay output isn't labelled, so it is sent to the first output file _out1.mp4_, regardless of the presence of the "-map" option. The aresample filter is sent the first unused audio stream, that of _A.avi_. Since this filter output is also unlabelled, it too is mapped to the first output file. The presence of "-an" only suppresses automatic or manual stream selection of audio streams, not outputs sent from filtergraphs. Both these mapped streams shall be ordered before the mapped stream in _out1.mp4_. The video, audio and subtitle streams mapped to "out2.mkv" are entirely determined by automatic stream selection. _out3.mkv_ consists of the cloned video output from the hue filter and the first audio stream from _B.mp4_. ## OPTIONS All the numerical options, if not specified otherwise, accept a string representing a number as input, which may be followed by one of the SI unit prefixes, for example: 'K', 'M', or 'G'. If 'i' is appended to the SI unit prefix, the complete prefix will be interpreted as a unit prefix for binary multiples, which are based on powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit prefix multiplies the value by 8. This allows using, for example: 'KB', 'MiB', 'G' and 'B' as number suffixes. Options which do not take arguments are boolean options, and set the corresponding value to true. They can be set to false by prefixing the option name with "no". For example using "-nofoo" will set the boolean option with name "foo" to false. ### Stream specifiers Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers are used to precisely specify which stream(s) a given option belongs to. A stream specifier is a string generally appended to the option name and separated from it by a colon. E.g. "-codec:a:1 ac3" contains the "a:1" stream specifier, which matches the second audio stream. Therefore, it would select the ac3 codec for the second audio stream. A stream specifier can match several streams, so that the option is applied to all of them. E.g. the stream specifier in "-b:a 128k" matches all audio streams. An empty stream specifier matches all streams. For example, "-codec copy" or "-codec: copy" would copy all the streams without reencoding. Possible forms of stream specifiers are: _stream_index_ Matches the stream with this index. E.g. "-threads:1 4" would set the thread count for the second stream to 4. If _stream_index_ is used as an additional stream specifier (see below), then it selects stream number _stream_index_ from the matching streams. Stream numbering is based on the order of the streams as detected by libavformat except when a program ID is also specified. In this case it is based on the ordering of the streams in the program. _stream_type_**[:_**additional_stream_specifier_**]** _stream_type_ is one of following: 'v' or 'V' for video, 'a' for audio, 's' for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video streams, 'V' only matches video streams which are not attached pictures, video thumbnails or cover arts. If _additional_stream_specifier_ is used, then it matches streams which both have this type and match the _additional_stream_specifier_. Otherwise, it matches all streams of the specified type. **p:_**program_id_**[:_**additional_stream_specifier_**]** Matches streams which are in the program with the id _program_id_. If _additional_stream_specifier_ is used, then it matches streams which both are part of the program and match the _additional_stream_specifier_. **#_**stream_id_ **or i:_**stream_id_ Match the stream by stream id (e.g. PID in MPEG-TS container). **m:_**key_**[:_**value_**]** Matches streams with the metadata tag _key_ having the specified value. If _value_ is not given, matches streams that contain the given tag with any value. **u **Matches streams with usable configuration, the codec must be defined and the essential information such as video dimension or audio sample rate must be present. Note that in **ffmpeg**, matching by metadata will only work properly for input files. ### Generic options These options are shared amongst the ff* tools. ### -L ### -h, -?, -help, --help [_ Show help. An optional parameter may be specified to print help about a specific item. If no argument is specified, only basic (non advanced) tool options are shown. Possible values of _arg_ are: **long** Print advanced tool options in addition to the basic tool options. **full** Print complete list of options, including shared and private options for encoders, decoders, demuxers, muxers, filters, etc. **decoder=_**decoder_name_ Print detailed information about the decoder named _decoder_name_. Use the **-decoders** option to get a list of all decoders. **encoder=_**encoder_name_ Print detailed information about the encoder named _encoder_name_. Use the **-encoders** option to get a list of all encoders. **demuxer=_**demuxer_name_ Print detailed information about the demuxer named _demuxer_name_. Use the **-formats** option to get a list of all demuxers and muxers. **muxer=_**muxer_name_ Print detailed information about the muxer named _muxer_name_. Use the **-formats **option to get a list of all muxers and demuxers. **filter=_**filter_name_ Print detailed information about the filter named _filter_name_. Use the **-filters** option to get a list of all filters. **bsf=_**bitstream_filter_name_ Print detailed information about the bitstream filter named _bitstream_filter_name_. Use the **-bsfs **option to get a list of all bitstream filters. **protocol=_**protocol_name_ Print detailed information about the protocol named _protocol_name_. Use the **-protocols **option to get a list of all protocols. ### -version Show version. ### -buildconf Show the build configuration, one option per line. ### -formats Show available formats (including devices). ### -demuxers Show available demuxers. ### -muxers Show available muxers. ### -devices Show available devices. ### -codecs Show all codecs known to libavcodec. Note that the term 'codec' is used throughout this documentation as a shortcut for what is more correctly called a media bitstream format. ### -decoders Show available decoders. ### -encoders Show all available encoders. ### -bsfs Show available bitstream filters. ### -protocols Show available protocols. ### -filters Show available libavfilter filters. ### -pix_fmts Show available pixel formats. ### -sample_fmts Show available sample formats. ### -layouts Show channel names and standard channel layouts. ### -dispositions Show stream dispositions. ### -colors Show recognized color names. ### -sources _ Show autodetected sources of the input device. Some devices may provide system-dependent source names that cannot be autodetected. The returned list cannot be assumed to be always complete. ffmpeg -sources pulse,server=192.168.0.4 ### -sinks _ Show autodetected sinks of the output device. Some devices may provide system-dependent sink names that cannot be autodetected. The returned list cannot be assumed to be always complete. ffmpeg -sinks pulse,server=192.168.0.4 ### -loglevel [_ Set logging level and flags used by the library. The optional _flags_ prefix can consist of the following values: **repeat** Indicates that repeated log output should not be compressed to the first line and the "Last message repeated n times" line will be omitted. **level** Indicates that log output should add a "[level]" prefix to each message line. This can be used as an alternative to log coloring, e.g. when dumping the log to file. Flags can also be used alone by adding a '+'/'-' prefix to set/reset a single flag without affecting other _flags_ or changing _loglevel_. When setting both _flags_ and _loglevel_, a '+' separator is expected between the last _flags_ value and before _loglevel_. _loglevel_ is a string or a number containing one of the following values: **quiet, -8** Show nothing at all; be silent. **panic, 0** Only show fatal errors which could lead the process to crash, such as an assertion failure. This is not currently used for anything. **fatal, 8** Only show fatal errors. These are errors after which the process absolutely cannot continue. **error, 16** Show all errors, including ones which can be recovered from. **warning, 24** Show all warnings and errors. Any message related to possibly incorrect or unexpected events will be shown. **info, 32** Show informative messages during processing. This is in addition to warnings and errors. This is the default value. **verbose, 40** Same as "info", except more verbose. **debug, 48** Show everything, including debugging information. **trace, 56** For example to enable repeated log output, add the "level" prefix, and set _loglevel_ to "verbose": ffmpeg -loglevel repeat+level+verbose -i input output Another example that enables repeated log output without affecting current state of "level" prefix flag or _loglevel_: ffmpeg [...] -loglevel +repeat By default the program logs to stderr. If coloring is supported by the terminal, colors are used to mark errors and warnings. Log coloring can be disabled setting the environment variable **AV_LOG_FORCE_NOCOLOR**, or can be forced setting the environment variable **AV_LOG_FORCE_COLOR**. ### -report Dump full command line and log output to a file named "_program_-_YYYYMMDD_-_HHMMSS_.log" in the current directory. This file can be useful for bug reports. It also implies "-loglevel debug". Setting the environment variable **FFREPORT **to any value has the same effect. If the value is a ':'-separated key=value sequence, these options will affect the report; option values must be escaped if they contain special characters or the options delimiter ':' (see the ``Quoting and escaping'' section in the ffmpeg-utils manual). The following options are recognized: **file** set the file name to use for the report; %p is expanded to the name of the program, %t is expanded to a timestamp, "%%" is expanded to a plain "%" **level** set the log verbosity level using a numerical value (see "-loglevel"). For example, to output a report to a file named _ffreport.log_ using a log level of 32 (alias for log level "info"): FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output Errors in parsing the environment variable are not fatal, and will not appear in the report. ### -hide_banner Suppress printing banner. All FFmpeg tools will normally show a copyright notice, build options and library versions. This option can be used to suppress printing this information. ### -cpuflags flags (_ Allows setting and clearing cpu flags. This option is intended for testing. Do not use it unless you know what you're doing. ffmpeg -cpuflags -sse+mmx ... ffmpeg -cpuflags mmx ... ffmpeg -cpuflags 0 ... Possible flags for this option are: **x86** **mmx** **mmxext** **sse** **sse2** **sse2slow** **sse3** **sse3slow** **ssse3** **atom** **sse4.1** **sse4.2** **avx** **avx2** **xop** **fma3** **fma4** **3dnow** **3dnowext** **bmi1** **bmi2** **cmov** **ARM** **armv5te** **armv6** **armv6t2** **vfp** **vfpv3** **neon** **setend** **AArch64** **armv8** **vfp** **neon** **PowerPC** **altivec** **Specific Processors** **pentium2** **pentium3** **pentium4** **k6** **k62** **athlon** **athlonxp** **k8** ### -cpucount _ Override detection of CPU count. This option is intended for testing. Do not use it unless you know what you're doing. ffmpeg -cpucount 2 ### -max_alloc _ Set the maximum size limit for allocating a block on the heap by ffmpeg's family of malloc functions. Exercise **extreme caution **when using this option. Don't use if you do not understand the full consequence of doing so. Default is INT_MAX. ### AVOptions These options are provided directly by the libavformat, libavdevice and libavcodec libraries. To see the list of available AVOptions, use the **-help **option. They are separated into two categories: ### generic These options can be set for any container, codec or device. Generic options are listed under AVFormatContext options for containers/devices and under AVCodecContext options for codecs. ### private These options are specific to the given container, device or codec. Private options are listed under their corresponding containers/devices/codecs. For example to write an ID3v2.3 header instead of a default ID3v2.4 to an MP3 file, use the **id3v2_version **private option of the MP3 muxer: ffmpeg -i input.flac -id3v2_version 3 out.mp3 All codec AVOptions are per-stream, and thus a stream specifier should be attached to them: ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4 In the above example, a multichannel audio stream is mapped twice for output. The first instance is encoded with codec ac3 and bitrate 640k. The second instance is downmixed to 2 channels and encoded with codec aac. A bitrate of 128k is specified for it using absolute index of the output stream. Note: the **-nooption **syntax cannot be used for boolean AVOptions, use **-option 0**/**-option 1**. Note: the old undocumented way of specifying per-stream AVOptions by prepending v/a/s to the options name is now obsolete and will be removed soon. ### Main options ### -f _ Force input or output file format. The format is normally auto detected for input files and guessed from the file extension for output files, so this option is not needed in most cases. ### -i _ input file url ### -y (_ Overwrite output files without asking. ### -n (_ Do not overwrite output files, and exit immediately if a specified output file already exists. ### -stream_loop _ Set number of times input stream shall be looped. Loop 0 means no loop, loop -1 means infinite loop. ### -recast_media (_ Allow forcing a decoder of a different media type than the one detected or designated by the demuxer. Useful for decoding media data muxed as data streams. ### -c[:_ ### -codec[:_ Select an encoder (when used before an output file) or a decoder (when used before an input file) for one or more streams. _codec_ is the name of a decoder/encoder or a special value "copy" (output only) to indicate that the stream is not to be re-encoded. For example ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT encodes all video streams with libx264 and copies all audio streams. For each stream, the last matching "c" option is applied, so ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT will copy all the streams except the second video, which will be encoded with libx264, and the 138th audio, which will be encoded with libvorbis. ### -t _ When used as an input option (before "-i"), limit the _duration_ of data read from the input file. When used as an output option (before an output url), stop writing the output after its duration reaches _duration_. _duration_ must be a time duration specification, see **the Time duration section in the** [**ffmpeg-utils(1)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown) manual**. -to and -t are mutually exclusive and -t has priority. ### -to _ Stop writing the output or reading the input at _position_. _position_ must be a time duration specification, see **the Time duration section in the [ffmpeg-utils(1)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown) manual**. -to and -t are mutually exclusive and -t has priority. ### -fs _ Set the file size limit, expressed in bytes. No further chunk of bytes is written after the limit is exceeded. The size of the output file is slightly more than the requested file size. ### -ss _ When used as an input option (before "-i"), seeks in this input file to _position_. Note that in most formats it is not possible to seek exactly, so **ffmpeg **will seek to the closest seek point before _position_. When transcoding and **-accurate_seek **is enabled (the default), this extra segment between the seek point and _position_ will be decoded and discarded. When doing stream copy or when **-noaccurate_seek **is used, it will be preserved. When used as an output option (before an output url), decodes but discards input until the timestamps reach _position_. _position_ must be a time duration specification, see **the Time duration section in the** [**ffmpeg-utils(1)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown) manual**. ### -sseof _ Like the "-ss" option but relative to the "end of file". That is negative values are earlier in the file, 0 is at EOF. ### -isync _ Assign an input as a sync source. This will take the difference between the start times of the target and reference inputs and offset the timestamps of the target file by that difference. The source timestamps of the two inputs should derive from the same clock source for expected results. If "copyts" is set then "start_at_zero" must also be set. If either of the inputs has no starting timestamp then no sync adjustment is made. Acceptable values are those that refer to a valid ffmpeg input index. If the sync reference is the target index itself or _-1_, then no adjustment is made to target timestamps. A sync reference may not itself be synced to any other input. Default value is _-1_. ### -itsoffset _ Set the input time offset. _offset_ must be a time duration specification, see **the Time duration section in the** [**ffmpeg-utils(1)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown) manual**. The offset is added to the timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by the time duration specified in _offset_. ### -itsscale _ Rescale input timestamps. _scale_ should be a floating point number. ### -timestamp _ Set the recording timestamp in the container. _date_ must be a date specification, see **the Date section in the [ffmpeg-utils(1)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown) manual**. ### -metadata[:metadata_specifier] _ Set a metadata key/value pair. An optional _metadata_specifier_ may be given to set metadata on streams, chapters or programs. See "-map_metadata" documentation for details. This option overrides metadata set with "-map_metadata". It is also possible to delete metadata by using an empty value. For example, for setting the title in the output file: ffmpeg -i in.avi -metadata title="my title" out.flv To set the language of the first audio stream: ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT ### -disposition[:stream_specifier] _ Sets the disposition for a stream. By default, the disposition is copied from the input stream, unless the output stream this option applies to is fed by a complex filtergraph - in that case the disposition is unset by default. _value_ is a sequence of items separated by '+' or '-'. The first item may also be prefixed with '+' or '-', in which case this option modifies the default value. Otherwise (the first item is not prefixed) this options overrides the default value. A '+' prefix adds the given disposition, '-' removes it. It is also possible to clear the disposition by setting it to 0. If no "-disposition" options were specified for an output file, ffmpeg will automatically set the 'default' disposition on the first stream of each type, when there are multiple streams of this type in the output file and no stream of that type is already marked as default. The "-dispositions" option lists the known dispositions. For example, to make the second audio stream the default stream: ffmpeg -i in.mkv -c copy -disposition:a:1 default out.mkv To make the second subtitle stream the default stream and remove the default disposition from the first subtitle stream: ffmpeg -i in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv To add an embedded cover/thumbnail: ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4 Not all muxers support embedded thumbnails, and those who do, only support a few formats, like JPEG or PNG. ### -program [title=_ Creates a program with the specified _title_, _program_num_ and adds the specified _stream_(s) to it. ### -target _ Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50"). _type_ may be prefixed with "pal-", "ntsc-" or "film-" to use the corresponding standard. All the format options (bitrate, codecs, buffer sizes) are then set automatically. You can just type: ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg Nevertheless you can specify additional options as long as you know they do not conflict with the standard, as in: ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg The parameters set for each target are as follows. **VCD** : -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324 -s 352x288 -r 25 -codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680 -ar 44100 -ac 2 -codec:a mp2 -b:a 224k : -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324 -s 352x240 -r 30000/1001 -codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680 -ar 44100 -ac 2 -codec:a mp2 -b:a 224k : -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324 -s 352x240 -r 24000/1001 -codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680 -ar 44100 -ac 2 -codec:a mp2 -b:a 224k **SVCD** : -f svcd -packetsize 2324 -s 480x576 -pix_fmt yuv420p -r 25 -codec:v mpeg2video -g 15 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1 -ar 44100 -codec:a mp2 -b:a 224k : -f svcd -packetsize 2324 -s 480x480 -pix_fmt yuv420p -r 30000/1001 -codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1 -ar 44100 -codec:a mp2 -b:a 224k : -f svcd -packetsize 2324 -s 480x480 -pix_fmt yuv420p -r 24000/1001 -codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1 -ar 44100 -codec:a mp2 -b:a 224k **DVD** : -f dvd -muxrate 10080k -packetsize 2048 -s 720x576 -pix_fmt yuv420p -r 25 -codec:v mpeg2video -g 15 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008 -ar 48000 -codec:a ac3 -b:a 448k : -f dvd -muxrate 10080k -packetsize 2048 -s 720x480 -pix_fmt yuv420p -r 30000/1001 -codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008 -ar 48000 -codec:a ac3 -b:a 448k : -f dvd -muxrate 10080k -packetsize 2048 -s 720x480 -pix_fmt yuv420p -r 24000/1001 -codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008 -ar 48000 -codec:a ac3 -b:a 448k **DV** : -f dv -s 720x576 -pix_fmt yuv420p -r 25 -ar 48000 -ac 2 : -f dv -s 720x480 -pix_fmt yuv411p -r 30000/1001 -ar 48000 -ac 2 : -f dv -s 720x480 -pix_fmt yuv411p -r 24000/1001 -ar 48000 -ac 2 The "dv50" target is identical to the "dv" target except that the pixel format set is "yuv422p" for all three standards. Any user-set value for a parameter above will override the target preset value. In that case, the output may not comply with the target standard. ### -dn (_ As an input option, blocks all data streams of a file from being filtered or being automatically selected or mapped for any output. See "-discard" option to disable streams individually. As an output option, disables data recording i.e. automatic selection or mapping of any data stream. For full manual control see the "-map" option. ### -dframes _ Set the number of data frames to output. This is an obsolete alias for "-frames:d", which you should use instead. ### -frames[:_ Stop writing to the stream after _framecount_ frames. ### -q[:_ ### -qscale[:_ Use fixed quality scale (VBR). The meaning of _q_/_qscale_ is codec-dependent. If _qscale_ is used without a _stream_specifier_ then it applies only to the video stream, this is to maintain compatibility with previous behavior and as specifying the same codec specific value to 2 different codecs that is audio and video generally is not what is intended when no stream_specifier is used. ### -filter[:_ Create the filtergraph specified by _filtergraph_ and use it to filter the stream. _filtergraph_ is a description of the filtergraph to apply to the stream, and must have a single input and a single output of the same type of the stream. In the filtergraph, the input is associated to the label "in", and the output to the label "out". See the ffmpeg- filters manual for more information about the filtergraph syntax. See the **-filter_complex option **if you want to create filtergraphs with multiple inputs and/or outputs. ### -filter_script[:_ This option is similar to **-filter**, the only difference is that its argument is the name of the file from which a filtergraph description is to be read. ### -reinit_filter[:_ This boolean option determines if the filtergraph(s) to which this stream is fed gets reinitialized when input frame parameters change mid-stream. This option is enabled by default as most video and all audio filters cannot handle deviation in input frame properties. Upon reinitialization, existing filter state is lost, like e.g. the frame count "n" reference available in some filters. Any frames buffered at time of reinitialization are lost. The properties where a change triggers reinitialization are, for video, frame resolution or pixel format; for audio, sample format, sample rate, channel count or channel layout. ### -filter_threads _ Defines how many threads are used to process a filter pipeline. Each pipeline will produce a thread pool with this many threads available for parallel processing. The default is the number of available CPUs. ### -pre[:_ Specify the preset for matching stream(s). ### -stats (_ Print encoding progress/statistics. It is on by default, to explicitly disable it you need to specify "-nostats". ### -stats_period _ Set period at which encoding progress/statistics are updated. Default is 0.5 seconds. ### -progress _ Send program-friendly progress information to _url_. Progress information is written periodically and at the end of the encoding process. It is made of "_key_=_value_" lines. _key_ consists of only alphanumeric characters. The last key of a sequence of progress information is always "progress". The update period is set using "-stats_period". ### -stdin Enable interaction on standard input. On by default unless standard input is used as an input. To explicitly disable interaction you need to specify "-nostdin". Disabling interaction on standard input is useful, for example, if ffmpeg is in the background process group. Roughly the same result can be achieved with "ffmpeg ... < /dev/null" but it requires a shell. ### -debug_ts (_ Print timestamp information. It is off by default. This option is mostly useful for testing and debugging purposes, and the output format may change from one version to another, so it should not be employed by portable scripts. See also the option "-fdebug ts". ### -attach _ Add an attachment to the output file. This is supported by a few formats like Matroska for e.g. fonts used in rendering subtitles. Attachments are implemented as a specific type of stream, so this option will add a new stream to the file. It is then possible to use per-stream options on this stream in the usual way. Attachment streams created with this option will be created after all the other streams (i.e. those created with "-map" or automatic mappings). Note that for Matroska you also have to set the mimetype metadata tag: ffmpeg -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv (assuming that the attachment stream will be third in the output file). ### -dump_attachment[:_ Extract the matching attachment stream into a file named _filename_. If _filename_ is empty, then the value of the "filename" metadata tag will be used. E.g. to extract the first attachment to a file named 'out.ttf': ffmpeg -dump_attachment:t:0 out.ttf -i INPUT To extract all attachments to files determined by the "filename" tag: ffmpeg -dump_attachment:t "" -i INPUT Technical note -- attachments are implemented as codec extradata, so this option can actually be used to extract extradata from any stream, not just attachments. ### Video Options ### -vframes _ Set the number of video frames to output. This is an obsolete alias for "-frames:v", which you should use instead. ### -r[:_ Set frame rate (Hz value, fraction or abbreviation). As an input option, ignore any timestamps stored in the file and instead generate timestamps assuming constant frame rate _fps_. This is not the same as the **-framerate** option used for some input formats like image2 or v4l2 (it used to be the same in older versions of FFmpeg). If in doubt use **-framerate **instead of the input option **-r**. As an output option: **video encoding** Duplicate or drop frames right before encoding them to achieve constant output frame rate _fps_. **video streamcopy** Indicate to the muxer that _fps_ is the stream frame rate. No data is dropped or duplicated in this case. This may produce invalid files if _fps_ does not match the actual stream frame rate as determined by packet timestamps. See also the "setts" bitstream filter. ### -fpsmax[:_ Set maximum frame rate (Hz value, fraction or abbreviation). Clamps output frame rate when output framerate is auto-set and is higher than this value. Useful in batch processing or when input framerate is wrongly detected as very high. It cannot be set together with "-r". It is ignored during streamcopy. ### -s[:_ Set frame size. As an input option, this is a shortcut for the **video_size **private option, recognized by some demuxers for which the frame size is either not stored in the file or is configurable -- e.g. raw video or video grabbers. As an output option, this inserts the "scale" video filter to the _end_ of the corresponding filtergraph. Please use the "scale" filter directly to insert it at the beginning or some other place. The format is **wxh **(default - same as source). ### -aspect[:_ Set the video display aspect ratio specified by _aspect_. _aspect_ can be a floating point number string, or a string of the form _num_:_den_, where _num_ and _den_ are the numerator and denominator of the aspect ratio. For example "4:3", "16:9", "1.3333", and "1.7777" are valid argument values. If used together with **-vcodec copy**, it will affect the aspect ratio stored at container level, but not the aspect ratio stored in encoded frames, if it exists. ### -display_rotation[:_ Set video rotation metadata. _rotation_ is a decimal number specifying the amount in degree by which the video should be rotated counter-clockwise before being displayed. This option overrides the rotation/display transform metadata stored in the file, if any. When the video is being transcoded (rather than copied) and "-autorotate" is enabled, the video will be rotated at the filtering stage. Otherwise, the metadata will be written into the output file if the muxer supports it. If the "-display_hflip" and/or "-display_vflip" options are given, they are applied after the rotation specified by this option. ### -display_hflip[:_ Set whether on display the image should be horizontally flipped. See the "-display_rotation" option for more details. ### -display_vflip[:_ Set whether on display the image should be vertically flipped. See the "-display_rotation" option for more details. ### -vn (_ As an input option, blocks all video streams of a file from being filtered or being automatically selected or mapped for any output. See "-discard" option to disable streams individually. As an output option, disables video recording i.e. automatic selection or mapping of any video stream. For full manual control see the "-map" option. ### -vcodec _ Set the video codec. This is an alias for "-codec:v". ### -pass[:_ Select the pass number (1 or 2). It is used to do two-pass video encoding. The statistics of the video are recorded in the first pass into a log file (see also the option -passlogfile), and in the second pass that log file is used to generate the video at the exact requested bitrate. On pass 1, you may just deactivate audio and set output to null, examples for Windows and Unix: ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null ### -passlogfile[:_ Set two-pass log file name prefix to _prefix_, the default file name prefix is ``ffmpeg2pass''. The complete file name will be _PREFIX-N.log_, where N is a number specific to the output stream ### -vf _ Create the filtergraph specified by _filtergraph_ and use it to filter the stream. This is an alias for "-filter:v", see the **-filter option**. ### -autorotate Automatically rotate the video according to file metadata. Enabled by default, use **-noautorotate **to disable it. ### -autoscale Automatically scale the video according to the resolution of first frame. Enabled by default, use **-noautoscale **to disable it. When autoscale is disabled, all output frames of filter graph might not be in the same resolution and may be inadequate for some encoder/muxer. Therefore, it is not recommended to disable it unless you really know what you are doing. Disable autoscale at your own risk. ### Advanced Video options ### -pix_fmt[:_ Set pixel format. Use "-pix_fmts" to show all the supported pixel formats. If the selected pixel format can not be selected, ffmpeg will print a warning and select the best pixel format supported by the encoder. If _pix_fmt_ is prefixed by a "+", ffmpeg will exit with an error if the requested pixel format can not be selected, and automatic conversions inside filtergraphs are disabled. If _pix_fmt_ is a single "+", ffmpeg selects the same pixel format as the input (or graph output) and automatic conversions are disabled. ### -sws_flags _ Set default flags for the libswscale library. These flags are used by automatically inserted "scale" filters and those within simple filtergraphs, if not overridden within the filtergraph definition. See the **ffmpeg-scaler manual **for a list of scaler options. ### -rc_override[:_ Rate control override for specific intervals, formatted as "int,int,int" list separated with slashes. Two first values are the beginning and end frame numbers, last one is quantizer to use if positive, or quality factor if negative. ### -psnr Calculate PSNR of compressed frames. This option is deprecated, pass the PSNR flag to the encoder instead, using "-flags +psnr". ### -vstats Dump video coding statistics to _vstats_HHMMSS.log_. See the **vstats file format **section for the format description. ### -vstats_file _ Dump video coding statistics to _file_. See the **vstats file format **section for the format description. ### -vstats_version _ Specify which version of the vstats format to use. Default is 2. See the **vstats file** **format **section for the format description. ### -vtag _ Force video tag/fourcc. This is an alias for "-tag:v". ### -vbsf _ Deprecated see -bsf ### -force_key_frames[:_ ### -force_key_frames[:_ ### -force_key_frames[:_ _force_key_frames_ can take arguments of the following form: _time_**[,_**time_**...]** If the argument consists of timestamps, ffmpeg will round the specified times to the nearest output timestamp as per the encoder time base and force a keyframe at the first frame having timestamp equal or greater than the computed timestamp. Note that if the encoder time base is too coarse, then the keyframes may be forced on frames with timestamps lower than the specified time. The default encoder time base is the inverse of the output framerate but may be set otherwise via "-enc_time_base". If one of the times is ""chapters"[_delta_]", it is expanded into the time of the beginning of all chapters in the file, shifted by _delta_, expressed as a time in seconds. This option can be useful to ensure that a seek point is present at a chapter mark or any other designated place in the output file. For example, to insert a key frame at 5 minutes, plus key frames 0.1 second before the beginning of every chapter: -force_key_frames 0:05:00,chapters-0.1 **expr:_**expr_ If the argument is prefixed with "expr:", the string _expr_ is interpreted like an expression and is evaluated for each frame. A key frame is forced in case the evaluation is non-zero. The expression in _expr_ can contain the following constants: **n **the number of current processed frame, starting from 0 **n_forced** the number of forced frames **prev_forced_n** the number of the previous forced frame, it is "NAN" when no keyframe was forced yet **prev_forced_t** the time of the previous forced frame, it is "NAN" when no keyframe was forced yet **t **the time of the current processed frame For example to force a key frame every 5 seconds, you can specify: -force_key_frames expr:gte(t,n_forced*5) To force a key frame 5 seconds after the time of the last forced one, starting from second 13: -force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5)) **source** If the argument is "source", ffmpeg will force a key frame if the current frame being encoded is marked as a key frame in its source. In cases where this particular source frame has to be dropped, enforce the next available frame to become a key frame instead. Note that forcing too many keyframes is very harmful for the lookahead algorithms of certain encoders: using fixed-GOP options or similar would be more efficient. ### -copyinkf[:_ When doing stream copy, copy also non-key frames found at the beginning. ### -init_hw_device _ Initialise a new hardware device of type _type_ called _name_, using the given device parameters. If no name is specified it will receive a default name of the form "_type_%d". The meaning of _device_ and the following arguments depends on the device type: **cuda** _device_ is the number of the CUDA device. The following options are recognized: **primary_ctx** If set to 1, uses the primary device context instead of creating a new one. Examples: _-init_hw_device_ _cuda:1_ Choose the second device on the system. _-init_hw_device_ _cuda:0,primary_ctx=1_ Choose the first device and use the primary device context. **dxva2** _device_ is the number of the Direct3D 9 display adapter. **d3d11va** _device_ is the number of the Direct3D 11 display adapter. **vaapi** _device_ is either an X11 display name, a DRM render node or a DirectX adapter index. If not specified, it will attempt to open the default X11 display (_$DISPLAY_) and then the first DRM render node (_/dev/dri/renderD128_), or the default DirectX adapter on Windows. **vdpau** _device_ is an X11 display name. If not specified, it will attempt to open the default X11 display (_$DISPLAY_). **qsv _**device_ selects a value in **MFX_IMPL_***. Allowed values are: **auto** **sw** **hw** **auto_any** **hw_any** **hw2** **hw3** **hw4** If not specified, **auto_any **is used. (Note that it may be easier to achieve the desired result for QSV by creating the platform-appropriate subdevice (**dxva2 **or **d3d11va **or **vaapi**) and then deriving a QSV device from that.) Alternatively, **child_device_type **helps to choose platform-appropriate subdevice type. On Windows **d3d11va **is used as default subdevice type. Examples: _-init_hw_device_ _qsv:hw,child_device_type=d3d11va_ Choose the GPU subdevice with type **d3d11va **and create QSV device with **MFX_IMPL_HARDWARE**. _-init_hw_device_ _qsv:hw,child_device_type=dxva2_ Choose the GPU subdevice with type **dxva2 **and create QSV device with **MFX_IMPL_HARDWARE**. **opencl** _device_ selects the platform and device as _platform_index.device_index_. The set of devices can also be filtered using the key-value pairs to find only devices matching particular platform or device strings. The strings usable as filters are: **platform_profile** **platform_version** **platform_name** **platform_vendor** **platform_extensions** **device_name** **device_vendor** **driver_version** **device_version** **device_profile** **device_extensions** **device_type** The indices and filters must together uniquely select a device. Examples: _-init_hw_device_ _opencl:0.1_ Choose the second device on the first platform. _-init_hw_device_ _opencl:,device_name=Foo9000_ Choose the device with a name containing the string _Foo9000_. _-init_hw_device_ _opencl:1,device_type=gpu,device_extensions=cl_khr_fp16_ Choose the GPU device on the second platform supporting the _cl_khr_fp16_ extension. **vulkan** If _device_ is an integer, it selects the device by its index in a system-dependent list of devices. If _device_ is any other string, it selects the first device with a name containing that string as a substring. The following options are recognized: **debug** If set to 1, enables the validation layer, if installed. **linear_images** If set to 1, images allocated by the hwcontext will be linear and locally mappable. **instance_extensions** A plus separated list of additional instance extensions to enable. **device_extensions** A plus separated list of additional device extensions to enable. Examples: _-init_hw_device_ _vulkan:1_ Choose the second device on the system. _-init_hw_device_ _vulkan:RADV_ Choose the first device with a name containing the string _RADV_. _-init_hw_device_ _vulkan:0,instance_extensions=VK_KHR_wayland_surface+VK_KHR_xcb_surface_ Choose the first device and enable the Wayland and XCB instance extensions. ### -init_hw_device _ Initialise a new hardware device of type _type_ called _name_, deriving it from the existing device with the name _source_. ### -init_hw_device list List all hardware device types supported in this build of ffmpeg. ### -filter_hw_device _ Pass the hardware device called _name_ to all filters in any filter graph. This can be used to set the device to upload to with the "hwupload" filter, or the device to map to with the "hwmap" filter. Other filters may also make use of this parameter when they require a hardware device. Note that this is typically only required when the input is not already in hardware frames - when it is, filters will derive the device they require from the context of the frames they receive as input. This is a global setting, so all filters will receive the same device. ### -hwaccel[:_ Use hardware acceleration to decode the matching stream(s). The allowed values of _hwaccel_ are: **none** Do not use any hardware acceleration (the default). **auto** Automatically select the hardware acceleration method. **vdpau** Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration. **dxva2** Use DXVA2 (DirectX Video Acceleration) hardware acceleration. **d3d11va** Use D3D11VA (DirectX Video Acceleration) hardware acceleration. **vaapi** Use VAAPI (Video Acceleration API) hardware acceleration. **qsv **Use the Intel QuickSync Video acceleration for video transcoding. Unlike most other values, this option does not enable accelerated decoding (that is used automatically whenever a qsv decoder is selected), but accelerated transcoding, without copying the frames into the system memory. For it to work, both the decoder and the encoder must support QSV acceleration and no filters must be used. This option has no effect if the selected hwaccel is not available or not supported by the chosen decoder. Note that most acceleration methods are intended for playback and will not be faster than software decoding on modern CPUs. Additionally, **ffmpeg **will usually need to copy the decoded frames from the GPU memory into the system memory, resulting in further performance loss. This option is thus mainly useful for testing. ### -hwaccel_device[:_ Select a device to use for hardware acceleration. This option only makes sense when the **-hwaccel **option is also specified. It can either refer to an existing device created with **-init_hw_device **by name, or it can create a new device as if **-init_hw_device _**type_:_hwaccel_device_ were called immediately before. ### -hwaccels List all hardware acceleration components enabled in this build of ffmpeg. Actual runtime availability depends on the hardware and its suitable driver being installed. ### -fix_sub_duration_heartbeat[:_ Set a specific output video stream as the heartbeat stream according to which to split and push through currently in-progress subtitle upon receipt of a random access packet. This lowers the latency of subtitles for which the end packet or the following subtitle has not yet been received. As a drawback, this will most likely lead to duplication of subtitle events in order to cover the full duration, so when dealing with use cases where latency of when the subtitle event is passed on to output is not relevant this option should not be utilized. Requires **-fix_sub_duration **to be set for the relevant input subtitle stream for this to have any effect, as well as for the input subtitle stream having to be directly mapped to the same output in which the heartbeat stream resides. ### Audio Options ### -aframes _ Set the number of audio frames to output. This is an obsolete alias for "-frames:a", which you should use instead. ### -ar[:_ Set the audio sampling frequency. For output streams it is set by default to the frequency of the corresponding input stream. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options. ### -aq _ Set the audio quality (codec-specific, VBR). This is an alias for -q:a. ### -ac[:_ Set the number of audio channels. For output streams it is set by default to the number of input audio channels. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options. ### -an (_ As an input option, blocks all audio streams of a file from being filtered or being automatically selected or mapped for any output. See "-discard" option to disable streams individually. As an output option, disables audio recording i.e. automatic selection or mapping of any audio stream. For full manual control see the "-map" option. ### -acodec _ Set the audio codec. This is an alias for "-codec:a". ### -sample_fmt[:_ Set the audio sample format. Use "-sample_fmts" to get a list of supported sample formats. ### -af _ Create the filtergraph specified by _filtergraph_ and use it to filter the stream. This is an alias for "-filter:a", see the **-filter option**. ### Advanced Audio options ### -atag _ Force audio tag/fourcc. This is an alias for "-tag:a". ### -absf _ Deprecated, see -bsf ### -guess_layout_max _ If some input channel layout is not known, try to guess only if it corresponds to at most the specified number of channels. For example, 2 tells to **ffmpeg **to recognize 1 channel as mono and 2 channels as stereo but not 6 channels as 5.1. The default is to always try to guess. Use 0 to disable all guessing. ### Subtitle options ### -scodec _ Set the subtitle codec. This is an alias for "-codec:s". ### -sn (_ As an input option, blocks all subtitle streams of a file from being filtered or being automatically selected or mapped for any output. See "-discard" option to disable streams individually. As an output option, disables subtitle recording i.e. automatic selection or mapping of any subtitle stream. For full manual control see the "-map" option. ### -sbsf _ Deprecated, see -bsf ### Advanced Subtitle options ### -fix_sub_duration Fix subtitles durations. For each subtitle, wait for the next packet in the same stream and adjust the duration of the first to avoid overlap. This is necessary with some subtitles codecs, especially DVB subtitles, because the duration in the original packet is only a rough estimate and the end is actually marked by an empty subtitle frame. Failing to use this option when necessary can result in exaggerated durations or muxing failures due to non-monotonic timestamps. Note that this option will delay the output of all data until the next subtitle packet is decoded: it may increase memory consumption and latency a lot. ### -canvas_size _ Set the size of the canvas used to render subtitles. ### Advanced options ### -map [-]_ Create one or more streams in the output file. This option has two forms for specifying the data source(s): the first selects one or more streams from some input file (specified with "-i"), the second takes an output from some complex filtergraph (specified with "-filter_complex" or "-filter_complex_script"). In the first form, an output stream is created for every stream from the input file with the index _input_file_id_. If _stream_specifier_ is given, only those streams that match the specifier are used (see the **Stream specifiers **section for the _stream_specifier_ syntax). A "-" character before the stream identifier creates a "negative" mapping. It disables matching streams from already created mappings. A trailing "?" after the stream index will allow the map to be optional: if the map matches no streams the map will be ignored instead of failing. Note the map will still fail if an invalid input file index is used; such as if the map refers to a non-existent input. An alternative _[linklabel]_ form will map outputs from complex filter graphs (see the **-filter_complex **option) to the output file. _linklabel_ must correspond to a defined output link label in the graph. This option may be specified multiple times, each adding more streams to the output file. Any given input stream may also be mapped any number of times as a source for different output streams, e.g. in order to use different encoding options and/or filters. The streams are created in the output in the same order in which the "-map" options are given on the commandline. Using this option disables the default mappings for this output file. Examples: _map_ _everything_ To map ALL streams from the first input file to output ffmpeg -i INPUT -map 0 output _select_ _specific_ _stream_ If you have two audio streams in the first input file, these streams are identified by _0:0_ and _0:1_. You can use "-map" to select which streams to place in an output file. For example: ffmpeg -i INPUT -map 0:1 out.wav will map the second input stream in _INPUT_ to the (single) output stream in _out.wav_. _create_ _multiple_ _streams_ To select the stream with index 2 from input file _a.mov_ (specified by the identifier _0:2_), and stream with index 6 from input _b.mov_ (specified by the identifier _1:6_), and copy them to the output file _out.mov_: ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov _create_ _multiple_ _streams_ _2_ To select all video and the third audio stream from an input file: ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT _negative_ _map_ To map all the streams except the second audio, use negative mappings ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT _optional_ _map_ To map the video and audio streams from the first input, and using the trailing "?", ignore the audio mapping if no audio streams exist in the first input: ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT _map_ _by_ _language_ To pick the English audio stream: ffmpeg -i INPUT -map 0:m:language:eng OUTPUT ### -ignore_unknown Ignore input streams with unknown type instead of failing if copying such streams is attempted. ### -copy_unknown Allow input streams with unknown type to be copied instead of failing if copying such streams is attempted. ### -map_channel **[_**input_file_id_**._**stream_specifier_**._**channel_id_**|-1][?][:_**output_file_id_**._**stream_specifier_**]** This option is deprecated and will be removed. It can be replaced by the _pan_ filter. In some cases it may be easier to use some combination of the _channelsplit_, _channelmap_, or _amerge_ filters. Map an audio channel from a given input to an output. If _output_file_id_._stream_specifier_ is not set, the audio channel will be mapped on all the audio streams. Using "-1" instead of _input_file_id_._stream_specifier_._channel_id_ will map a muted channel. A trailing "?" will allow the map_channel to be optional: if the map_channel matches no channel the map_channel will be ignored instead of failing. For example, assuming _INPUT_ is a stereo audio file, you can switch the two audio channels with the following command: ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT If you want to mute the first channel and keep the second: ffmpeg -i INPUT -map_channel -1 -map_channel 0.0.1 OUTPUT The order of the "-map_channel" option specifies the order of the channels in the output stream. The output channel layout is guessed from the number of channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac" in combination of "-map_channel" makes the channel gain levels to be updated if input and output channel layouts don't match (for instance two "-map_channel" options and "-ac 6"). You can also extract each channel of an input to specific outputs; the following command extracts two channels of the _INPUT_ audio stream (file 0, stream 0) to the respective _OUTPUT_CH0_ and _OUTPUT_CH1_ outputs: ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1 The following example splits the channels of a stereo input into two separate streams, which are put into the same output file: ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg Note that currently each output stream can only contain channels from a single input stream; you can't for example use "-map_channel" to pick multiple input audio channels contained in different streams (from the same or different files) and merge them into a single output stream. It is therefore not currently possible, for example, to turn two separate mono streams into a single stereo stream. However splitting a stereo stream into two single channel mono streams is possible. If you need this feature, a possible workaround is to use the _amerge_ filter. For example, if you need to merge a media (here _input.mkv_) with 2 mono audio streams into one single stereo channel audio stream (and keep the video stream), you can use the following command: ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv To map the first two audio channels from the first input, and using the trailing "?", ignore the audio channel mapping if the first input is mono instead of stereo: ffmpeg -i INPUT -map_channel 0.0.0 -map_channel 0.0.1? OUTPUT ### -map_metadata[:_ Set metadata information of the next output file from _infile_. Note that those are file indices (zero-based), not filenames. Optional _metadata_spec_in/out_ parameters specify, which metadata to copy. A metadata specifier can have the following forms: _g_ global metadata, i.e. metadata that applies to the whole file _s_**[:_**stream_spec_**]** per-stream metadata. _stream_spec_ is a stream specifier as described in the **Stream** **specifiers **chapter. In an input metadata specifier, the first matching stream is copied from. In an output metadata specifier, all matching streams are copied to. _c_**:_**chapter_index_ per-chapter metadata. _chapter_index_ is the zero-based chapter index. _p_**:_**program_index_ per-program metadata. _program_index_ is the zero-based program index. If metadata specifier is omitted, it defaults to global. By default, global metadata is copied from the first input file, per-stream and per- chapter metadata is copied along with streams/chapters. These default mappings are disabled by creating any mapping of the relevant type. A negative file index can be used to create a dummy mapping that just disables automatic copying. For example to copy metadata from the first stream of the input file to global metadata of the output file: ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3 To do the reverse, i.e. copy global metadata to all audio streams: ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv Note that simple 0 would work as well in this example, since global metadata is assumed by default. ### -map_chapters _ Copy chapters from input file with index _input_file_index_ to the next output file. If no chapter mapping is specified, then chapters are copied from the first input file with at least one chapter. Use a negative file index to disable any chapter copying. ### -benchmark (_ Show benchmarking information at the end of an encode. Shows real, system and user time used and maximum memory consumption. Maximum memory consumption is not supported on all systems, it will usually display as 0 if not supported. ### -benchmark_all (_ Show benchmarking information during the encode. Shows real, system and user time used in various steps (audio/video encode/decode). ### -timelimit _ Exit after ffmpeg has been running for _duration_ seconds in CPU user time. ### -dump (_ Dump each input packet to stderr. ### -hex (_ When dumping packets, also dump the payload. ### -readrate _ Limit input read speed. Its value is a floating-point positive number which represents the maximum duration of media, in seconds, that should be ingested in one second of wallclock time. Default value is zero and represents no imposed limitation on speed of ingestion. Value 1 represents real-time speed and is equivalent to "-re". Mainly used to simulate a capture device or live input stream (e.g. when reading from a file). Should not be used with a low value when input is an actual capture device or live stream as it may cause packet loss. It is useful for when flow speed of output packets is important, such as live streaming. ### -re (_ Read input at native frame rate. This is equivalent to setting "-readrate 1". ### -readrate_initial_burst _ Set an initial read burst time, in seconds, after which **-re/-readrate **will be enforced. ### -vsync _ ### -fps_mode[:_ Set video sync method / framerate mode. vsync is applied to all output video streams but can be overridden for a stream by setting fps_mode. vsync is deprecated and will be removed in the future. For compatibility reasons some of the values for vsync can be specified as numbers (shown in parentheses in the following table). **passthrough (0)** Each frame is passed with its timestamp from the demuxer to the muxer. **cfr (1)** Frames will be duplicated and dropped to achieve exactly the requested constant frame rate. **vfr (2)** Frames are passed through with their timestamp or dropped so as to prevent 2 frames from having the same timestamp. **drop** As passthrough but destroys all timestamps, making the muxer generate fresh timestamps based on frame-rate. **auto (-1)** Chooses between cfr and vfr depending on muxer capabilities. This is the default method. Note that the timestamps may be further modified by the muxer, after this. For example, in the case that the format option **avoid_negative_ts **is enabled. With -map you can select from which stream the timestamps should be taken. You can leave either video or audio unchanged and sync the remaining stream(s) to the unchanged one. ### -frame_drop_threshold _ Frame drop threshold, which specifies how much behind video frames can be before they are dropped. In frame rate units, so 1.0 is one frame. The default is -1.1. One possible usecase is to avoid framedrops in case of noisy timestamps or to increase frame drop precision in case of exact timestamps. ### -apad _ Pad the output audio stream(s). This is the same as applying "-af apad". Argument is a string of filter parameters composed the same as with the "apad" filter. "-shortest" must be set for this output for the option to take effect. ### -copyts Do not process input timestamps, but keep their values without trying to sanitize them. In particular, do not remove the initial start time offset value. Note that, depending on the **vsync **option or on specific muxer processing (e.g. in case the format option **avoid_negative_ts **is enabled) the output timestamps may mismatch with the input timestamps even when this option is selected. ### -start_at_zero When used with **copyts**, shift input timestamps so they start at zero. This means that using e.g. "-ss 50" will make output timestamps start at 50 seconds, regardless of what timestamp the input file started at. ### -copytb _ Specify how to set the encoder timebase when stream copying. _mode_ is an integer numeric value, and can assume one of the following values: **1 **Use the demuxer timebase. The time base is copied to the output encoder from the corresponding input demuxer. This is sometimes required to avoid non monotonically increasing timestamps when copying video streams with variable frame rate. **0 **Use the decoder timebase. The time base is copied to the output encoder from the corresponding input decoder. **-1 **Try to make the choice automatically, in order to generate a sane output. Default value is -1. ### -enc_time_base[:_ Set the encoder timebase. _timebase_ can assume one of the following values: **0 **Assign a default value according to the media type. For video - use 1/framerate, for audio - use 1/samplerate. **demux** Use the timebase from the demuxer. **filter** Use the timebase from the filtergraph. **a positive number** Use the provided number as the timebase. This field can be provided as a ratio of two integers (e.g. 1:24, 1:48000) or as a decimal number (e.g. 0.04166, 2.0833e-5) Default value is 0. ### -bitexact (_ Enable bitexact mode for (de)muxer and (de/en)coder ### -shortest (_ Finish encoding when the shortest output stream ends. Note that this option may require buffering frames, which introduces extra latency. The maximum amount of this latency may be controlled with the "-shortest_buf_duration" option. ### -shortest_buf_duration _ The "-shortest" option may require buffering potentially large amounts of data when at least one of the streams is "sparse" (i.e. has large gaps between frames – this is typically the case for subtitles). This option controls the maximum duration of buffered frames in seconds. Larger values may allow the "-shortest" option to produce more accurate results, but increase memory use and latency. The default value is 10 seconds. ### -dts_delta_threshold _ Timestamp discontinuity delta threshold, expressed as a decimal number of seconds. The timestamp discontinuity correction enabled by this option is only applied to input formats accepting timestamp discontinuity (for which the "AV_FMT_DISCONT" flag is enabled), e.g. MPEG-TS and HLS, and is automatically disabled when employing the "-copy_ts" option (unless wrapping is detected). If a timestamp discontinuity is detected whose absolute value is greater than _threshold_, ffmpeg will remove the discontinuity by decreasing/increasing the current DTS and PTS by the corresponding delta value. The default value is 10. ### -dts_error_threshold _ Timestamp error delta threshold, expressed as a decimal number of seconds. The timestamp correction enabled by this option is only applied to input formats not accepting timestamp discontinuity (for which the "AV_FMT_DISCONT" flag is not enabled). If a timestamp discontinuity is detected whose absolute value is greater than _threshold_, ffmpeg will drop the PTS/DTS timestamp value. The default value is "3600*30" (30 hours), which is arbitrarily picked and quite conservative. ### -muxdelay _ Set the maximum demux-decode delay. ### -muxpreload _ Set the initial demux-decode delay. ### -streamid _ Assign a new stream-id value to an output stream. This option should be specified prior to the output filename to which it applies. For the situation where multiple output files exist, a streamid may be reassigned to a different value. For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for an output mpegts file: ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts ### -bsf[:_ Set bitstream filters for matching streams. _bitstream_filters_ is a comma-separated list of bitstream filters. Use the "-bsfs" option to get the list of bitstream filters. ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264 ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt ### -tag[:_ Force a tag/fourcc for matching streams. ### -timecode _ Specify Timecode for writing. _SEP_ is ':' for non drop timecode and ';' (or '.') for drop. ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg ### -filter_complex _ Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or outputs. For simple graphs -- those with one input and one output of the same type -- see the **-filter **options. _filtergraph_ is a description of the filtergraph, as described in the ``Filtergraph syntax'' section of the ffmpeg-filters manual. Input link labels must refer to input streams using the "[file_index:stream_specifier]" syntax (i.e. the same as **-map **uses). If _stream_specifier_ matches multiple streams, the first one will be used. An unlabeled input will be connected to the first unused input stream of the matching type. Output link labels are referred to with **-map**. Unlabeled outputs are added to the first output file. Note that with this option it is possible to use only lavfi sources without normal input files. For example, to overlay an image over video ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map '[out]' out.mkv Here "[0:v]" refers to the first video stream in the first input file, which is linked to the first (main) input of the overlay filter. Similarly the first video stream in the second input is linked to the second (overlay) input of overlay. Assuming there is only one video stream in each input file, we can omit input labels, so the above is equivalent to ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map '[out]' out.mkv Furthermore we can omit the output label and the single output from the filter graph will be added to the output file automatically, so we can simply write ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv As a special exception, you can use a bitmap subtitle stream as input: it will be converted into a video with the same size as the largest video in the file, or 720x576 if no video is present. Note that this is an experimental and temporary solution. It will be removed once libavfilter has proper support for subtitles. For example, to hardcode subtitles on top of a DVB-T recording stored in MPEG-TS format, delaying the subtitles by 1 second: ffmpeg -i input.ts -filter_complex \ '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \ -sn -map '#0x2dc' output.mkv (0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video, audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too) To generate 5 seconds of pure red video using lavfi "color" source: ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv ### -filter_complex_threads _ Defines how many threads are used to process a filter_complex graph. Similar to filter_threads but used for "-filter_complex" graphs only. The default is the number of available CPUs. ### -lavfi _ Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or outputs. Equivalent to **-filter_complex**. ### -filter_complex_script _ This option is similar to **-filter_complex**, the only difference is that its argument is the name of the file from which a complex filtergraph description is to be read. ### -accurate_seek (_ This option enables or disables accurate seeking in input files with the **-ss **option. It is enabled by default, so seeking is accurate when transcoding. Use **-noaccurate_seek **to disable it, which may be useful e.g. when copying some streams and transcoding the others. ### -seek_timestamp (_ This option enables or disables seeking by timestamp in input files with the **-ss **option. It is disabled by default. If enabled, the argument to the **-ss **option is considered an actual timestamp, and is not offset by the start time of the file. This matters only for files which do not start from timestamp 0, such as transport streams. ### -thread_queue_size _ For input, this option sets the maximum number of queued packets when reading from the file or device. With low latency / high rate live streams, packets may be discarded if they are not read in a timely manner; setting this value can force ffmpeg to use a separate input thread and read packets as soon as they arrive. By default ffmpeg only does this if multiple inputs are specified. For output, this option specified the maximum number of packets that may be queued to each muxing thread. ### -sdp_file _ Print sdp information for an output stream to _file_. This allows dumping sdp information when at least one output isn't an rtp stream. (Requires at least one of the output formats to be rtp). ### -discard (_ Allows discarding specific streams or frames from streams. Any input stream can be fully discarded, using value "all" whereas selective discarding of frames from a stream occurs at the demuxer and is not supported by all demuxers. **none** Discard no frame. **default** Default, which discards no frames. **noref** Discard all non-reference frames. **bidir** Discard all bidirectional frames. **nokey** Discard all frames excepts keyframes. **all **Discard all frames. ### -abort_on _ Stop and abort on various conditions. The following flags are available: **empty_output** No packets were passed to the muxer, the output is empty. **empty_output_stream** No packets were passed to the muxer in some of the output streams. ### -max_error_rate (_ Set fraction of decoding frame failures across all inputs which when crossed ffmpeg will return exit code 69. Crossing this threshold does not terminate processing. Range is a floating-point number between 0 to 1. Default is 2/3. ### -xerror (_ Stop and exit on error ### -max_muxing_queue_size _ When transcoding audio and/or video streams, ffmpeg will not begin writing into the output until it has one packet for each such stream. While waiting for that to happen, packets for other streams are buffered. This option sets the size of this buffer, in packets, for the matching output stream. The default value of this option should be high enough for most uses, so only touch this option if you are sure that you need it. ### -muxing_queue_data_threshold _ This is a minimum threshold until which the muxing queue size is not taken into account. Defaults to 50 megabytes per stream, and is based on the overall size of packets passed to the muxer. ### -auto_conversion_filters (_ Enable automatically inserting format conversion filters in all filter graphs, including those defined by **-vf**, **-af**, **-filter_complex **and **-lavfi**. If filter format negotiation requires a conversion, the initialization of the filters will fail. Conversions can still be performed by inserting the relevant conversion filter (scale, aresample) in the graph. On by default, to explicitly disable it you need to specify "-noauto_conversion_filters". ### -bits_per_raw_sample[:_ Declare the number of bits per raw sample in the given output stream to be _value_. Note that this option sets the information provided to the encoder/muxer, it does not change the stream to conform to this value. Setting values that do not match the stream properties may result in encoding failures or invalid output files. ### -stats_enc_pre[:_ ### -stats_enc_post[:_ ### -stats_mux_pre[:_ Write per-frame encoding information about the matching streams into the file given by _path_. **-stats_enc_pre **writes information about raw video or audio frames right before they are sent for encoding, while **-stats_enc_post **writes information about encoded packets as they are received from the encoder. **-stats_mux_pre **writes information about packets just as they are about to be sent to the muxer. Every frame or packet produces one line in the specified file. The format of this line is controlled by **-stats_enc_pre_fmt **/ **-stats_enc_post_fmt **/ **-stats_mux_pre_fmt**. When stats for multiple streams are written into a single file, the lines corresponding to different streams will be interleaved. The precise order of this interleaving is not specified and not guaranteed to remain stable between different invocations of the program, even with the same options. ### -stats_enc_pre_fmt[:_ ### -stats_enc_post_fmt[:_ ### -stats_mux_pre_fmt[:_ Specify the format for the lines written with **-stats_enc_pre **/ **-stats_enc_post **/ **-stats_mux_pre**. _format_spec_ is a string that may contain directives of the form _{fmt}_. _format_spec_ is backslash-escaped --- use \{, \}, and \\ to write a literal {, }, or \, respectively, into the output. The directives given with _fmt_ may be one of the following: **fidx** Index of the output file. **sidx** Index of the output stream in the file. **n **Frame number. Pre-encoding: number of frames sent to the encoder so far. Post- encoding: number of packets received from the encoder so far. Muxing: number of packets submitted to the muxer for this stream so far. **ni **Input frame number. Index of the input frame (i.e. output by a decoder) that corresponds to this output frame or packet. -1 if unavailable. **tb **Timebase in which this frame/packet's timestamps are expressed, as a rational number _num/den_. Note that encoder and muxer may use different timebases. **tbi **Timebase for _ptsi_, as a rational number _num/den_. Available when _ptsi_ is available, _0/1_ otherwise. **pts **Presentation timestamp of the frame or packet, as an integer. Should be multiplied by the timebase to compute presentation time. **ptsi** Presentation timestamp of the input frame (see _ni_), as an integer. Should be multiplied by _tbi_ to compute presentation time. Printed as (2^63 - 1 = 9223372036854775807) when not available. **t **Presentation time of the frame or packet, as a decimal number. Equal to _pts_ multiplied by _tb_. **ti **Presentation time of the input frame (see _ni_), as a decimal number. Equal to _ptsi_ multiplied by _tbi_. Printed as inf when not available. **dts (_**packet_**)** Decoding timestamp of the packet, as an integer. Should be multiplied by the timebase to compute presentation time. **dt (_**packet_**)** Decoding time of the frame or packet, as a decimal number. Equal to _dts_ multiplied by _tb_. **sn (_**frame,audio_**)** Number of audio samples sent to the encoder so far. **samp (_**frame,audio_**)** Number of audio samples in the frame. **size (_**packet_**)** Size of the encoded packet in bytes. **br (_**packet_**)** Current bitrate in bits per second. Post-encoding only. **abr (_**packet_**)** Average bitrate for the whole stream so far, in bits per second, -1 if it cannot be determined at this point. Post-encoding only. Directives tagged with _packet_ may only be used with **-stats_enc_post_fmt **and **-stats_mux_pre_fmt**. Directives tagged with _frame_ may only be used with **-stats_enc_pre_fmt**. Directives tagged with _audio_ may only be used with audio streams. The default format strings are: **pre-encoding** {fidx} {sidx} {n} {t} **post-encoding** {fidx} {sidx} {n} {t} In the future, new items may be added to the end of the default formatting strings. Users who depend on the format staying exactly the same, should prescribe it manually. Note that stats for different streams written into the same file may have different formats. ### Preset files A preset file contains a sequence of _option_=_value_ pairs, one for each line, specifying a sequence of options which would be awkward to specify on the command line. Lines starting with the hash ('#') character are ignored and are used to provide comments. Check the _presets_ directory in the FFmpeg source tree for examples. There are two types of preset files: ffpreset and avpreset files. _ffpreset_ _files_ ffpreset files are specified with the "vpre", "apre", "spre", and "fpre" options. The "fpre" option takes the filename of the preset instead of a preset name as input and can be used for any kind of codec. For the "vpre", "apre", and "spre" options, the options specified in a preset file are applied to the currently selected codec of the same type as the preset option. The argument passed to the "vpre", "apre", and "spre" preset options identifies the preset file to use according to the following rules: First ffmpeg searches for a file named _arg_.ffpreset in the directories _$FFMPEG_DATADIR_ (if set), and _$HOME/.ffmpeg_, and in the datadir defined at configuration time (usually _PREFIX/share/ffmpeg_) or in a _ffpresets_ folder along the executable on win32, in that order. For example, if the argument is "libvpx-1080p", it will search for the file _libvpx-1080p.ffpreset_. If no such file is found, then ffmpeg will search for a file named _codec_name_-_arg_.ffpreset in the above-mentioned directories, where _codec_name_ is the name of the codec to which the preset file options will be applied. For example, if you select the video codec with "-vcodec libvpx" and use "-vpre 1080p", then it will search for the file _libvpx-1080p.ffpreset_. _avpreset_ _files_ avpreset files are specified with the "pre" option. They work similar to ffpreset files, but they only allow encoder- specific options. Therefore, an _option_=_value_ pair specifying an encoder cannot be used. When the "pre" option is specified, ffmpeg will look for files with the suffix .avpreset in the directories _$AVCONV_DATADIR_ (if set), and _$HOME/.avconv_, and in the datadir defined at configuration time (usually _PREFIX/share/ffmpeg_), in that order. First ffmpeg searches for a file named _codec_name_-_arg_.avpreset in the above-mentioned directories, where _codec_name_ is the name of the codec to which the preset file options will be applied. For example, if you select the video codec with "-vcodec libvpx" and use "-pre 1080p", then it will search for the file _libvpx-1080p.avpreset_. If no such file is found, then ffmpeg will search for a file named _arg_.avpreset in the same directories. ### vstats file format The "-vstats" and "-vstats_file" options enable generation of a file containing statistics about the generated video outputs. The "-vstats_version" option controls the format version of the generated file. With version 1 the format is: frame= q= PSNR= f_size= s_size= kB time= br= kbits/s avg_br= kbits/s With version 2 the format is: out= st= frame= q= f PSNR= f_size= s_size= kB time= br= kbits/s avg_br= kbits/s The value corresponding to each key is described below: ### avg_br average bitrate expressed in Kbits/s **br **bitrate expressed in Kbits/s ### frame number of encoded frame **out **out file index **PSNR** Peak Signal to Noise Ratio **q **quality of the frame ### f_size encoded packet size expressed as number of bytes ### s_size stream size expressed in KiB **st **out file stream index ### time time of the packet ### type picture type See also the **-stats_enc options **for an alternative way to show encoding statistics. ## EXAMPLES ### Video and Audio grabbing If you specify the input format and device then ffmpeg can grab video and audio directly. ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg Or with an ALSA audio source (mono input, card id 1) instead of OSS: ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg Note that you must activate the right video source and channel before launching ffmpeg with any TV viewer such as <****> by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer. ### X11 grabbing Grab the X11 display with ffmpeg via ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg 0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg 0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. 10 is the x-offset and 20 the y-offset for the grabbing. ### Video and Audio file format conversion Any supported file format and protocol can serve as input to ffmpeg: Examples: • You can use YUV files as input: ffmpeg -i /tmp/test%d.Y /tmp/out.mpg It will use the files: /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V, /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc... The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the **-s **option if ffmpeg cannot guess it. • You can input from a raw YUV420P file: ffmpeg -i /tmp/test.yuv /tmp/out.avi test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed by the U and V planes at half vertical and horizontal resolution. • You can output to a raw YUV420P file: ffmpeg -i mydivx.avi hugefile.yuv • You can set several input files and output files: ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg. • You can also do audio and video conversions at the same time: ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2 Converts a.wav to MPEG audio at 22050 Hz sample rate. • You can encode to several formats at the same time and define a mapping from input stream to output streams: ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2 Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map file:index' specifies which input stream is used for each output stream, in the order of the definition of output streams. • You can transcode decrypted VOBs: ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi This is a typical DVD ripping example; the input is a VOB file, the output an AVI file with MPEG-4 video and MP3 audio. Note that in this command we use B-frames so the MPEG-4 stream is DivX5 compatible, and GOP size is 300 which means one intra frame every 10 seconds for 29.97fps input video. Furthermore, the audio stream is MP3-encoded so you need to enable LAME support by passing "--enable-libmp3lame" to configure. The mapping is particularly useful for DVD transcoding to get the desired audio language. NOTE: To see the supported input formats, use "ffmpeg -demuxers". • You can extract images from a video, or create a video from many images: For extracting images from a video: ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg This will extract one video frame per second from the video and will output them in files named _foo-001.jpeg_, _foo-002.jpeg_, etc. Images will be rescaled to fit the new WxH values. If you want to extract just a limited number of frames, you can use the above command in combination with the "-frames:v" or "-t" option, or in combination with -ss to start extracting from a certain point in time. For creating a video from many images: ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi The syntax "foo-%03d.jpeg" specifies to use a decimal number composed of three digits padded with zeroes to express the sequence number. It is the same syntax supported by the C printf function, but only formats accepting a normal integer are suitable. When importing an image sequence, -i also supports expanding shell-like wildcard patterns (globbing) internally, by selecting the image2-specific "-pattern_type glob" option. For example, for creating a video from filenames matching the glob pattern "foo-*.jpeg": ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi • You can put many streams of the same type in the output: ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut The resulting output file _test12.nut_ will contain the first four streams from the input files in reverse order. • To force CBR video output: ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v • The four options lmin, lmax, mblmin and mblmax use 'lambda' units, but you may use the QP2LAMBDA constant to easily convert from 'q' units: ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext ## SYNTAX This section documents the syntax and formats employed by the FFmpeg libraries and tools. ### Quoting and escaping FFmpeg adopts the following quoting and escaping mechanism, unless explicitly specified. The following rules are applied: • **' **and **\ **are special characters (respectively used for quoting and escaping). In addition to them, there might be other special characters depending on the specific syntax where the escaping and quoting are employed. • A special character is escaped by prefixing it with a **\**. • All characters enclosed between **'' **are included literally in the parsed string. The quote character **' **itself cannot be quoted, so you may need to close the quote and escape it. • Leading and trailing whitespaces, unless escaped or quoted, are removed from the parsed string. Note that you may need to add a second level of escaping when using the command line or a script, which depends on the syntax of the adopted shell language. The function "av_get_token" defined in _libavutil/avstring.h_ can be used to parse a token quoted or escaped according to the rules defined above. The tool _tools/ffescape_ in the FFmpeg source tree can be used to automatically quote or escape a string in a script. _Examples_ • Escape the string "Crime d'Amour" containing the "'" special character: Crime d\'Amour • The string above contains a quote, so the "'" needs to be escaped when quoting it: 'Crime d'\''Amour' • Include leading or trailing whitespaces using quoting: ' this string starts and ends with whitespaces ' • Escaping and quoting can be mixed together: ' The string '\'string\'' is a string ' • To include a literal **\ **you can use either escaping or quoting: 'c:\foo' can be written as c:\\foo ### Date The accepted syntax is: [(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z] now If the value is "now" it takes the current time. Time is local time unless Z is appended, in which case it is interpreted as UTC. If the year-month-day part is not specified it takes the current year-month-day. ### Time duration There are two accepted syntaxes for expressing time duration. [-][:]:[....] _HH_ expresses the number of hours, _MM_ the number of minutes for a maximum of 2 digits, and _SS_ the number of seconds for a maximum of 2 digits. The _m_ at the end expresses decimal value for _SS_. _or_ [-]+[....][s|ms|us] _S_ expresses the number of seconds, with the optional decimal part _m_. The optional literal suffixes **s**, **ms **or **us **indicate to interpret the value as seconds, milliseconds or microseconds, respectively. In both expressions, the optional **- **indicates negative duration. _Examples_ The following examples are all valid time duration: **55 **55 seconds **0.2 **0.2 seconds ### 200ms 200 milliseconds, that's 0.2s ### 200000us 200000 microseconds, that's 0.2s ### 12:03:45 12 hours, 03 minutes and 45 seconds ### 23.189 23.189 seconds ### Video size Specify the size of the sourced video, it may be a string of the form _width_x_height_, or the name of a size abbreviation. The following abbreviations are recognized: ### ntsc 720x480 **pal **720x576 ### qntsc 352x240 ### qpal 352x288 ### sntsc 640x480 ### spal 768x576 ### film 352x240 ### ntsc-film 352x240 ### sqcif 128x96 ### qcif 176x144 **cif **352x288 ### 4cif 704x576 ### 16cif 1408x1152 ### qqvga 160x120 ### qvga 320x240 **vga **640x480 ### svga 800x600 **xga **1024x768 ### uxga 1600x1200 ### qxga 2048x1536 ### sxga 1280x1024 ### qsxga 2560x2048 ### hsxga 5120x4096 ### wvga 852x480 ### wxga 1366x768 ### wsxga 1600x1024 ### wuxga 1920x1200 ### woxga 2560x1600 ### wqsxga 3200x2048 ### wquxga 3840x2400 ### whsxga 6400x4096 ### whuxga 7680x4800 **cga **320x200 **ega **640x350 ### hd480 852x480 ### hd720 1280x720 ### hd1080 1920x1080 **2k **2048x1080 ### 2kflat 1998x1080 ### 2kscope 2048x858 **4k **4096x2160 ### 4kflat 3996x2160 ### 4kscope 4096x1716 **nhd **640x360 ### hqvga 240x160 ### wqvga 400x240 ### fwqvga 432x240 ### hvga 480x320 **qhd **960x540 ### 2kdci 2048x1080 ### 4kdci 4096x2160 ### uhd2160 3840x2160 ### uhd4320 7680x4320 ### Video rate Specify the frame rate of a video, expressed as the number of frames generated per second. It has to be a string in the format _frame_rate_num_/_frame_rate_den_, an integer number, a float number or a valid video frame rate abbreviation. The following abbreviations are recognized: ### ntsc 30000/1001 **pal **25/1 ### qntsc 30000/1001 ### qpal 25/1 ### sntsc 30000/1001 ### spal 25/1 ### film 24/1 ### ntsc-film 24000/1001 ### Ratio A ratio can be expressed as an expression, or in the form _numerator_:_denominator_. Note that a ratio with infinite (1/0) or negative value is considered valid, so you should check on the returned value if you want to exclude those values. The undefined value can be expressed using the "0:0" string. ### Color It can be the name of a color as defined below (case insensitive match) or a "[0x|#]RRGGBB[AA]" sequence, possibly followed by @ and a string representing the alpha component. The alpha component may be a string composed by "0x" followed by an hexadecimal number or a decimal number between 0.0 and 1.0, which represents the opacity value (**0x00 **or **0.0 **means completely transparent, **0xff **or **1.0 **completely opaque). If the alpha component is not specified then **0xff **is assumed. The string **random **will result in a random color. The following names of colors are recognized: ### AliceBlue 0xF0F8FF ### AntiqueWhite 0xFAEBD7 ### Aqua 0x00FFFF ### Aquamarine 0x7FFFD4 ### Azure 0xF0FFFF ### Beige 0xF5F5DC ### Bisque 0xFFE4C4 ### Black 0x000000 ### BlanchedAlmond 0xFFEBCD ### Blue 0x0000FF ### BlueViolet 0x8A2BE2 ### Brown 0xA52A2A ### BurlyWood 0xDEB887 ### CadetBlue 0x5F9EA0 ### Chartreuse 0x7FFF00 ### Chocolate 0xD2691E ### Coral 0xFF7F50 ### CornflowerBlue 0x6495ED ### Cornsilk 0xFFF8DC ### Crimson 0xDC143C ### Cyan 0x00FFFF ### DarkBlue 0x00008B ### DarkCyan 0x008B8B ### DarkGoldenRod 0xB8860B ### DarkGray 0xA9A9A9 ### DarkGreen 0x006400 ### DarkKhaki 0xBDB76B ### DarkMagenta 0x8B008B ### DarkOliveGreen 0x556B2F ### Darkorange 0xFF8C00 ### DarkOrchid 0x9932CC ### DarkRed 0x8B0000 ### DarkSalmon 0xE9967A ### DarkSeaGreen 0x8FBC8F ### DarkSlateBlue 0x483D8B ### DarkSlateGray 0x2F4F4F ### DarkTurquoise 0x00CED1 ### DarkViolet 0x9400D3 ### DeepPink 0xFF1493 ### DeepSkyBlue 0x00BFFF ### DimGray 0x696969 ### DodgerBlue 0x1E90FF ### FireBrick 0xB22222 ### FloralWhite 0xFFFAF0 ### ForestGreen 0x228B22 ### Fuchsia 0xFF00FF ### Gainsboro 0xDCDCDC ### GhostWhite 0xF8F8FF ### Gold 0xFFD700 ### GoldenRod 0xDAA520 ### Gray 0x808080 ### Green 0x008000 ### GreenYellow 0xADFF2F ### HoneyDew 0xF0FFF0 ### HotPink 0xFF69B4 ### IndianRed 0xCD5C5C ### Indigo 0x4B0082 ### Ivory 0xFFFFF0 ### Khaki 0xF0E68C ### Lavender 0xE6E6FA ### LavenderBlush 0xFFF0F5 ### LawnGreen 0x7CFC00 ### LemonChiffon 0xFFFACD ### LightBlue 0xADD8E6 ### LightCoral 0xF08080 ### LightCyan 0xE0FFFF ### LightGoldenRodYellow 0xFAFAD2 ### LightGreen 0x90EE90 ### LightGrey 0xD3D3D3 ### LightPink 0xFFB6C1 ### LightSalmon 0xFFA07A ### LightSeaGreen 0x20B2AA ### LightSkyBlue 0x87CEFA ### LightSlateGray 0x778899 ### LightSteelBlue 0xB0C4DE ### LightYellow 0xFFFFE0 ### Lime 0x00FF00 ### LimeGreen 0x32CD32 ### Linen 0xFAF0E6 ### Magenta 0xFF00FF ### Maroon 0x800000 ### MediumAquaMarine 0x66CDAA ### MediumBlue 0x0000CD ### MediumOrchid 0xBA55D3 ### MediumPurple 0x9370D8 ### MediumSeaGreen 0x3CB371 ### MediumSlateBlue 0x7B68EE ### MediumSpringGreen 0x00FA9A ### MediumTurquoise 0x48D1CC ### MediumVioletRed 0xC71585 ### MidnightBlue 0x191970 ### MintCream 0xF5FFFA ### MistyRose 0xFFE4E1 ### Moccasin 0xFFE4B5 ### NavajoWhite 0xFFDEAD ### Navy 0x000080 ### OldLace 0xFDF5E6 ### Olive 0x808000 ### OliveDrab 0x6B8E23 ### Orange 0xFFA500 ### OrangeRed 0xFF4500 ### Orchid 0xDA70D6 ### PaleGoldenRod 0xEEE8AA ### PaleGreen 0x98FB98 ### PaleTurquoise 0xAFEEEE ### PaleVioletRed 0xD87093 ### PapayaWhip 0xFFEFD5 ### PeachPuff 0xFFDAB9 ### Peru 0xCD853F ### Pink 0xFFC0CB ### Plum 0xDDA0DD ### PowderBlue 0xB0E0E6 ### Purple 0x800080 **Red **0xFF0000 ### RosyBrown 0xBC8F8F ### RoyalBlue 0x4169E1 ### SaddleBrown 0x8B4513 ### Salmon 0xFA8072 ### SandyBrown 0xF4A460 ### SeaGreen 0x2E8B57 ### SeaShell 0xFFF5EE ### Sienna 0xA0522D ### Silver 0xC0C0C0 ### SkyBlue 0x87CEEB ### SlateBlue 0x6A5ACD ### SlateGray 0x708090 ### Snow 0xFFFAFA ### SpringGreen 0x00FF7F ### SteelBlue 0x4682B4 **Tan **0xD2B48C ### Teal 0x008080 ### Thistle 0xD8BFD8 ### Tomato 0xFF6347 ### Turquoise 0x40E0D0 ### Violet 0xEE82EE ### Wheat 0xF5DEB3 ### White 0xFFFFFF ### WhiteSmoke 0xF5F5F5 ### Yellow 0xFFFF00 ### YellowGreen 0x9ACD32 ### Channel Layout A channel layout specifies the spatial disposition of the channels in a multi-channel audio stream. To specify a channel layout, FFmpeg makes use of a special syntax. Individual channels are identified by an id, as given by the table below: **FL **front left **FR **front right **FC **front center **LFE **low frequency **BL **back left **BR **back right **FLC **front left-of-center **FRC **front right-of-center **BC **back center **SL **side left **SR **side right **TC **top center **TFL **top front left **TFC **top front center **TFR **top front right **TBL **top back left **TBC **top back center **TBR **top back right **DL **downmix left **DR **downmix right **WL **wide left **WR **wide right **SDL **surround direct left **SDR **surround direct right **LFE2** low frequency 2 Standard channel layout compositions can be specified by using the following identifiers: ### mono FC ### stereo FL+FR **2.1 **FL+FR+LFE **3.0 **FL+FR+FC ### 3.0(back) FL+FR+BC **4.0 **FL+FR+FC+BC ### quad FL+FR+BL+BR ### quad(side) FL+FR+SL+SR **3.1 **FL+FR+FC+LFE **5.0 **FL+FR+FC+BL+BR ### 5.0(side) FL+FR+FC+SL+SR **4.1 **FL+FR+FC+LFE+BC **5.1 **FL+FR+FC+LFE+BL+BR ### 5.1(side) FL+FR+FC+LFE+SL+SR **6.0 **FL+FR+FC+BC+SL+SR ### 6.0(front) FL+FR+FLC+FRC+SL+SR ### 3.1.2 FL+FR+FC+LFE+TFL+TFR ### hexagonal FL+FR+FC+BL+BR+BC **6.1 **FL+FR+FC+LFE+BC+SL+SR **6.1 **FL+FR+FC+LFE+BL+BR+BC ### 6.1(front) FL+FR+LFE+FLC+FRC+SL+SR **7.0 **FL+FR+FC+BL+BR+SL+SR ### 7.0(front) FL+FR+FC+FLC+FRC+SL+SR **7.1 **FL+FR+FC+LFE+BL+BR+SL+SR ### 7.1(wide) FL+FR+FC+LFE+BL+BR+FLC+FRC ### 7.1(wide-side) FL+FR+FC+LFE+FLC+FRC+SL+SR ### 5.1.2 FL+FR+FC+LFE+BL+BR+TFL+TFR ### octagonal FL+FR+FC+BL+BR+BC+SL+SR ### cube FL+FR+BL+BR+TFL+TFR+TBL+TBR ### 5.1.4 FL+FR+FC+LFE+BL+BR+TFL+TFR+TBL+TBR ### 7.1.2 FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR ### 7.1.4 FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBL+TBR ### hexadecagonal FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR ### downmix DL+DR ### 22.2 FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL+SR+TC+TFL+TFC+TFR+TBL+TBC+TBR+LFE2+TSL+TSR+BFC+BFL+BFR A custom channel layout can be specified as a sequence of terms, separated by '+'. Each term can be: • the name of a single channel (e.g. **FL**, **FR**, **FC**, **LFE**, etc.), each optionally containing a custom name after a '@', (e.g. **FL@Left**, **FR@Right**, **FC@Center**, **LFE@Low_Frequency**, etc.) A standard channel layout can be specified by the following: • the name of a single channel (e.g. **FL**, **FR**, **FC**, **LFE**, etc.) • the name of a standard channel layout (e.g. **mono**, **stereo**, **4.0**, **quad**, **5.0**, etc.) • a number of channels, in decimal, followed by 'c', yielding the default channel layout for that number of channels (see the function "av_channel_layout_default"). Note that not all channel counts have a default layout. • a number of channels, in decimal, followed by 'C', yielding an unknown channel layout with the specified number of channels. Note that not all channel layout specification strings support unknown channel layouts. • a channel layout mask, in hexadecimal starting with "0x" (see the "AV_CH_*" macros in _libavutil/channel_layout.h_. Before libavutil version 53 the trailing character "c" to specify a number of channels was optional, but now it is required, while a channel layout mask can also be specified as a decimal number (if and only if not followed by "c" or "C"). See also the function "av_channel_layout_from_string" defined in _libavutil/channel_layout.h_. ## EXPRESSION EVALUATION When evaluating an arithmetic expression, FFmpeg uses an internal formula evaluator, implemented through the _libavutil/eval.h_ interface. An expression may contain unary, binary operators, constants, and functions. Two expressions _expr1_ and _expr2_ can be combined to form another expression "_expr1_;_expr2_". _expr1_ and _expr2_ are evaluated in turn, and the new expression evaluates to the value of _expr2_. The following binary operators are available: "+", "-", "*", "/", "^". The following unary operators are available: "+", "-". The following functions are available: ### abs(x) Compute absolute value of _x_. ### acos(x) Compute arccosine of _x_. ### asin(x) Compute arcsine of _x_. ### atan(x) Compute arctangent of _x_. ### atan2(x, y) Compute principal value of the arc tangent of _y_/_x_. ### between(x, min, max) Return 1 if _x_ is greater than or equal to _min_ and lesser than or equal to _max_, 0 otherwise. ### bitand(x, y) ### bitor(x, y) Compute bitwise and/or operation on _x_ and _y_. The results of the evaluation of _x_ and _y_ are converted to integers before executing the bitwise operation. Note that both the conversion to integer and the conversion back to floating point can lose precision. Beware of unexpected results for large numbers (usually 2^53 and larger). ### ceil(expr) Round the value of expression _expr_ upwards to the nearest integer. For example, "ceil(1.5)" is "2.0". ### clip(x, min, max) Return the value of _x_ clipped between _min_ and _max_. ### cos(x) Compute cosine of _x_. ### cosh(x) Compute hyperbolic cosine of _x_. ### eq(x, y) Return 1 if _x_ and _y_ are equivalent, 0 otherwise. ### exp(x) Compute exponential of _x_ (with base "e", the Euler's number). ### floor(expr) Round the value of expression _expr_ downwards to the nearest integer. For example, "floor(-1.5)" is "-2.0". ### gauss(x) Compute Gauss function of _x_, corresponding to "exp(-x*x/2) / sqrt(2*PI)". ### gcd(x, y) Return the greatest common divisor of _x_ and _y_. If both _x_ and _y_ are 0 or either or both are less than zero then behavior is undefined. ### gt(x, y) Return 1 if _x_ is greater than _y_, 0 otherwise. ### gte(x, y) Return 1 if _x_ is greater than or equal to _y_, 0 otherwise. ### hypot(x, y) This function is similar to the C function with the same name; it returns "sqrt(_x_*_x_ + _y_*_y_)", the length of the hypotenuse of a right triangle with sides of length _x_ and _y_, or the distance of the point (_x_, _y_) from the origin. ### if(x, y) Evaluate _x_, and if the result is non-zero return the result of the evaluation of _y_, return 0 otherwise. ### if(x, y, z) Evaluate _x_, and if the result is non-zero return the evaluation result of _y_, otherwise the evaluation result of _z_. ### ifnot(x, y) Evaluate _x_, and if the result is zero return the result of the evaluation of _y_, return 0 otherwise. ### ifnot(x, y, z) Evaluate _x_, and if the result is zero return the evaluation result of _y_, otherwise the evaluation result of _z_. ### isinf(x) Return 1.0 if _x_ is +/-INFINITY, 0.0 otherwise. ### isnan(x) Return 1.0 if _x_ is NAN, 0.0 otherwise. ### ld(var) Load the value of the internal variable with number _var_, which was previously stored with st(_var_, _expr_). The function returns the loaded value. ### lerp(x, y, z) Return linear interpolation between _x_ and _y_ by amount of _z_. ### log(x) Compute natural logarithm of _x_. ### lt(x, y) Return 1 if _x_ is lesser than _y_, 0 otherwise. ### lte(x, y) Return 1 if _x_ is lesser than or equal to _y_, 0 otherwise. ### max(x, y) Return the maximum between _x_ and _y_. ### min(x, y) Return the minimum between _x_ and _y_. ### mod(x, y) Compute the remainder of division of _x_ by _y_. ### not(expr) Return 1.0 if _expr_ is zero, 0.0 otherwise. ### pow(x, y) Compute the power of _x_ elevated _y_, it is equivalent to "(_x_)^(_y_)". ### print(t) ### print(t, l) Print the value of expression _t_ with loglevel _l_. If _l_ is not specified then a default log level is used. Returns the value of the expression printed. Prints t with loglevel l ### random(x) Return a pseudo random value between 0.0 and 1.0. _x_ is the index of the internal variable which will be used to save the seed/state. ### root(expr, max) Find an input value for which the function represented by _expr_ with argument _**[ld**(0)](https://www.chedong.com/phpMan.php/man/ld/0/markdown)_ is 0 in the interval 0.._max_. The expression in _expr_ must denote a continuous function or the result is undefined. _**[ld**(0)](https://www.chedong.com/phpMan.php/man/ld/0/markdown)_ is used to represent the function input value, which means that the given expression will be evaluated multiple times with various input values that the expression can access through [ld(0)](https://www.chedong.com/phpMan.php/man/ld/0/markdown). When the expression evaluates to 0 then the corresponding input value will be returned. ### round(expr) Round the value of expression _expr_ to the nearest integer. For example, "round(1.5)" is "2.0". ### sgn(x) Compute sign of _x_. ### sin(x) Compute sine of _x_. ### sinh(x) Compute hyperbolic sine of _x_. ### sqrt(expr) Compute the square root of _expr_. This is equivalent to "(_expr_)^.5". ### squish(x) Compute expression "1/(1 + exp(4*x))". ### st(var, expr) Store the value of the expression _expr_ in an internal variable. _var_ specifies the number of the variable where to store the value, and it is a value ranging from 0 to 9. The function returns the value stored in the internal variable. Note, Variables are currently not shared between expressions. ### tan(x) Compute tangent of _x_. ### tanh(x) Compute hyperbolic tangent of _x_. ### taylor(expr, x) ### taylor(expr, x, id) Evaluate a Taylor series at _x_, given an expression representing the ld(id)-th derivative of a function at 0. When the series does not converge the result is undefined. _ld(id)_ is used to represent the derivative order in _expr_, which means that the given expression will be evaluated multiple times with various input values that the expression can access through ld(id). If _id_ is not specified then 0 is assumed. Note, when you have the derivatives at y instead of 0, "taylor(expr, x-y)" can be used. ### [time(0)](https://www.chedong.com/phpMan.php/man/time/0/markdown) Return the current (wallclock) time in seconds. ### trunc(expr) Round the value of expression _expr_ towards zero to the nearest integer. For example, "trunc(-1.5)" is "-1.0". ### while(cond, expr) Evaluate expression _expr_ while the expression _cond_ is non-zero, and returns the value of the last _expr_ evaluation, or NAN if _cond_ was always false. The following constants are available: **PI **area of the unit disc, approximately 3.14 **E [exp**(1)](https://www.chedong.com/phpMan.php/man/exp/1/markdown) (Euler's number), approximately 2.718 **PHI **golden ratio (1+[**sqrt**(5)](https://www.chedong.com/phpMan.php/man/sqrt/5/markdown))/2, approximately 1.618 Assuming that an expression is considered "true" if it has a non-zero value, note that: "*" works like AND "+" works like OR For example the construct: if (A AND B) then C is equivalent to: if(A*B, C) In your C code, you can extend the list of unary and binary functions, and define recognized constants, so that they are available for your expressions. The evaluator also recognizes the International System unit prefixes. If 'i' is appended after the prefix, binary prefixes are used, which are based on powers of 1024 instead of powers of 1000. The 'B' postfix multiplies the value by 8, and can be appended after a unit prefix or used alone. This allows using for example 'KB', 'MiB', 'G' and 'B' as number postfix. The list of available International System prefixes follows, with indication of the corresponding powers of 10 and of 2. **y **10^-24 / 2^-80 **z **10^-21 / 2^-70 **a **10^-18 / 2^-60 **f **10^-15 / 2^-50 **p **10^-12 / 2^-40 **n **10^-9 / 2^-30 **u **10^-6 / 2^-20 **m **10^-3 / 2^-10 **c **10^-2 **d **10^-1 **h **10^2 **k **10^3 / 2^10 **K **10^3 / 2^10 **M **10^6 / 2^20 **G **10^9 / 2^30 **T **10^12 / 2^40 **P **10^15 / 2^50 **E **10^18 / 2^60 **Z **10^21 / 2^70 **Y **10^24 / 2^80 ## CODEC OPTIONS libavcodec provides some generic global options, which can be set on all the encoders and decoders. In addition each codec may support so-called private options, which are specific for a given codec. Sometimes, a global option may only affect a specific kind of codec, and may be nonsensical or ignored by another, so you need to be aware of the meaning of the specified options. Also some options are meant only for decoding or encoding. Options may be set by specifying -_option_ _value_ in the FFmpeg tools, or by setting the value explicitly in the "AVCodecContext" options or using the _libavutil/opt.h_ API for programmatic use. The list of supported options follow: **b _**integer_ **(_**encoding,audio,video_**)** Set bitrate in bits/s. Default value is 200K. **ab _**integer_ **(_**encoding,audio_**)** Set audio bitrate (in bits/s). Default value is 128K. **bt _**integer_ **(_**encoding,video_**)** Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate tolerance specifies how far ratecontrol is willing to deviate from the target average bitrate value. This is not related to min/max bitrate. Lowering tolerance too much has an adverse effect on quality. **flags _**flags_ **(_**decoding/encoding,audio,video,subtitles_**)** Set generic flags. Possible values: **mv4 **Use four motion vector by macroblock (mpeg4). **qpel** Use 1/4 pel motion compensation. **loop** Use loop filter. **qscale** Use fixed qscale. **pass1** Use internal 2pass ratecontrol in first pass mode. **pass2** Use internal 2pass ratecontrol in second pass mode. **gray** Only decode/encode grayscale. **psnr** Set error[?] variables during encoding. **truncated** Input bitstream might be randomly truncated. **drop_changed** Don't output frames whose parameters differ from first decoded frame in stream. Error AVERROR_INPUT_CHANGED is returned when a frame is dropped. **ildct** Use interlaced DCT. **low_delay** Force low delay. **global_header** Place global headers in extradata instead of every keyframe. **bitexact** Only write platform-, build- and time-independent data. (except (I)DCT). This ensures that file and data checksums are reproducible and match between platforms. Its primary use is for regression testing. **aic **Apply H263 advanced intra coding / mpeg4 ac prediction. **ilme** Apply interlaced motion estimation. **cgop** Use closed gop. **output_corrupt** Output even potentially corrupted frames. **time_base _**rational_ _number_ Set codec time base. It is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented. For fixed-fps content, timebase should be "1 / frame_rate" and timestamp increments should be identically 1. **g _**integer_ **(_**encoding,video_**)** Set the group of picture (GOP) size. Default value is 12. **ar _**integer_ **(_**decoding/encoding,audio_**)** Set audio sampling rate (in Hz). **ac _**integer_ **(_**decoding/encoding,audio_**)** Set number of audio channels. **cutoff _**integer_ **(_**encoding,audio_**)** Set cutoff bandwidth. (Supported only by selected encoders, see their respective documentation sections.) **frame_size _**integer_ **(_**encoding,audio_**)** Set audio frame size. Each submitted frame except the last must contain exactly frame_size samples per channel. May be 0 when the codec has CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is not restricted. It is set by some decoders to indicate constant frame size. **frame_number _**integer_ Set the frame number. **delay _**integer_ **qcomp _**float_ **(_**encoding,video_**)** Set video quantizer scale compression (VBR). It is used as a constant in the ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0. **qblur _**float_ **(_**encoding,video_**)** Set video quantizer scale blur (VBR). **qmin _**integer_ **(_**encoding,video_**)** Set min video quantizer scale (VBR). Must be included between -1 and 69, default value is 2. **qmax _**integer_ **(_**encoding,video_**)** Set max video quantizer scale (VBR). Must be included between -1 and 1024, default value is 31. **qdiff _**integer_ **(_**encoding,video_**)** Set max difference between the quantizer scale (VBR). **bf _**integer_ **(_**encoding,video_**)** Set max number of B frames between non-B-frames. Must be an integer between -1 and 16. 0 means that B-frames are disabled. If a value of -1 is used, it will choose an automatic value depending on the encoder. Default value is 0. **b_qfactor _**float_ **(_**encoding,video_**)** Set qp factor between P and B frames. **codec_tag _**integer_ **bug _**flags_ **(_**decoding,video_**)** Workaround not auto detected encoder bugs. Possible values: **autodetect** **xvid_ilace** Xvid interlacing bug (autodetected if fourcc==XVIX) **ump4** (autodetected if fourcc==UMP4) **no_padding** padding bug (autodetected) **amv** **qpel_chroma** **std_qpel** old standard qpel (autodetected per fourcc/version) **qpel_chroma2** **direct_blocksize** direct-qpel-blocksize bug (autodetected per fourcc/version) **edge** edge padding bug (autodetected per fourcc/version) **hpel_chroma** **dc_clip** **ms **Workaround various bugs in microsoft broken decoders. **trunc** trancated frames **strict _**integer_ **(_**decoding/encoding,audio,video_**)** Specify how strictly to follow the standards. Possible values: **very** strictly conform to an older more strict version of the spec or reference software **strict** strictly conform to all the things in the spec no matter what consequences **normal** **unofficial** allow unofficial extensions **experimental** allow non standardized experimental things, experimental (unfinished/work in progress/not well tested) decoders and encoders. Note: experimental decoders can pose a security risk, do not use this for decoding untrusted input. **b_qoffset _**float_ **(_**encoding,video_**)** Set QP offset between P and B frames. **err_detect _**flags_ **(_**decoding,audio,video_**)** Set error detection flags. Possible values: **crccheck** verify embedded CRCs **bitstream** detect bitstream specification deviations **buffer** detect improper bitstream length **explode** abort decoding on minor error detection **ignore_err** ignore decoding errors, and continue decoding. This is useful if you want to analyze the content of a video and thus want everything to be decoded no matter what. This option will not result in a video that is pleasing to watch in case of errors. **careful** consider things that violate the spec and have not been seen in the wild as errors **compliant** consider all spec non compliancies as errors **aggressive** consider things that a sane encoder should not do as an error **has_b_frames _**integer_ **block_align _**integer_ **rc_override_count _**integer_ **maxrate _**integer_ **(_**encoding,audio,video_**)** Set max bitrate tolerance (in bits/s). Requires bufsize to be set. **minrate _**integer_ **(_**encoding,audio,video_**)** Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use elsewise. **bufsize _**integer_ **(_**encoding,audio,video_**)** Set ratecontrol buffer size (in bits). **i_qfactor _**float_ **(_**encoding,video_**)** Set QP factor between P and I frames. **i_qoffset _**float_ **(_**encoding,video_**)** Set QP offset between P and I frames. **dct _**integer_ **(_**encoding,video_**)** Set DCT algorithm. Possible values: **auto** autoselect a good one (default) **fastint** fast integer **int **accurate integer **mmx** **altivec** **faan** floating point AAN DCT **lumi_mask _**float_ **(_**encoding,video_**)** Compress bright areas stronger than medium ones. **tcplx_mask _**float_ **(_**encoding,video_**)** Set temporal complexity masking. **scplx_mask _**float_ **(_**encoding,video_**)** Set spatial complexity masking. **p_mask _**float_ **(_**encoding,video_**)** Set inter masking. **dark_mask _**float_ **(_**encoding,video_**)** Compress dark areas stronger than medium ones. **idct _**integer_ **(_**decoding/encoding,video_**)** Select IDCT implementation. Possible values: **auto** **int** **simple** **simplemmx** **simpleauto** Automatically pick a IDCT compatible with the simple one **arm** **altivec** **sh4** **simplearm** **simplearmv5te** **simplearmv6** **simpleneon** **xvid** **faani** floating point AAN IDCT **slice_count _**integer_ **ec _**flags_ **(_**decoding,video_**)** Set error concealment strategy. Possible values: **guess_mvs** iterative motion vector (MV) search (slow) **deblock** use strong deblock filter for damaged MBs **favor_inter** favor predicting from the previous frame instead of the current **bits_per_coded_sample _**integer_ **aspect _**rational_ _number_ **(_**encoding,video_**)** Set sample aspect ratio. **sar _**rational_ _number_ **(_**encoding,video_**)** Set sample aspect ratio. Alias to _aspect_. **debug _**flags_ **(_**decoding/encoding,audio,video,subtitles_**)** Print specific debug info. Possible values: **pict** picture info **rc **rate control **bitstream** **mb_type** macroblock (MB) type **qp **per-block quantization parameter (QP) **dct_coeff** **green_metadata** display complexity metadata for the upcoming frame, GoP or for a given duration. **skip** **startcode** **er **error recognition **mmco** memory management control operations (H.264) **bugs** **buffers** picture buffer allocations **thread_ops** threading operations **nomc** skip motion compensation **cmp _**integer_ **(_**encoding,video_**)** Set full pel me compare function. Possible values: **sad **sum of absolute differences, fast (default) **sse **sum of squared errors **satd** sum of absolute Hadamard transformed differences **dct **sum of absolute DCT transformed differences **psnr** sum of squared quantization errors (avoid, low quality) **bit **number of bits needed for the block **rd **rate distortion optimal, slow **zero** 0 **vsad** sum of absolute vertical differences **vsse** sum of squared vertical differences **nsse** noise preserving sum of squared differences **w53 **5/3 wavelet, only used in snow **w97 **9/7 wavelet, only used in snow **dctmax** **chroma** **subcmp _**integer_ **(_**encoding,video_**)** Set sub pel me compare function. Possible values: **sad **sum of absolute differences, fast (default) **sse **sum of squared errors **satd** sum of absolute Hadamard transformed differences **dct **sum of absolute DCT transformed differences **psnr** sum of squared quantization errors (avoid, low quality) **bit **number of bits needed for the block **rd **rate distortion optimal, slow **zero** 0 **vsad** sum of absolute vertical differences **vsse** sum of squared vertical differences **nsse** noise preserving sum of squared differences **w53 **5/3 wavelet, only used in snow **w97 **9/7 wavelet, only used in snow **dctmax** **chroma** **mbcmp _**integer_ **(_**encoding,video_**)** Set macroblock compare function. Possible values: **sad **sum of absolute differences, fast (default) **sse **sum of squared errors **satd** sum of absolute Hadamard transformed differences **dct **sum of absolute DCT transformed differences **psnr** sum of squared quantization errors (avoid, low quality) **bit **number of bits needed for the block **rd **rate distortion optimal, slow **zero** 0 **vsad** sum of absolute vertical differences **vsse** sum of squared vertical differences **nsse** noise preserving sum of squared differences **w53 **5/3 wavelet, only used in snow **w97 **9/7 wavelet, only used in snow **dctmax** **chroma** **ildctcmp _**integer_ **(_**encoding,video_**)** Set interlaced dct compare function. Possible values: **sad **sum of absolute differences, fast (default) **sse **sum of squared errors **satd** sum of absolute Hadamard transformed differences **dct **sum of absolute DCT transformed differences **psnr** sum of squared quantization errors (avoid, low quality) **bit **number of bits needed for the block **rd **rate distortion optimal, slow **zero** 0 **vsad** sum of absolute vertical differences **vsse** sum of squared vertical differences **nsse** noise preserving sum of squared differences **w53 **5/3 wavelet, only used in snow **w97 **9/7 wavelet, only used in snow **dctmax** **chroma** **dia_size _**integer_ **(_**encoding,video_**)** Set diamond type & size for motion estimation. **(1024, INT_MAX)** full motion estimation(slowest) **(768, 1024]** umh motion estimation **(512, 768]** hex motion estimation **(256, 512]** l2s diamond motion estimation **[2,256]** var diamond motion estimation **(-1, 2)** small diamond motion estimation **-1 **funny diamond motion estimation **(INT_MIN, -1)** sab diamond motion estimation **last_pred _**integer_ **(_**encoding,video_**)** Set amount of motion predictors from the previous frame. **precmp _**integer_ **(_**encoding,video_**)** Set pre motion estimation compare function. Possible values: **sad **sum of absolute differences, fast (default) **sse **sum of squared errors **satd** sum of absolute Hadamard transformed differences **dct **sum of absolute DCT transformed differences **psnr** sum of squared quantization errors (avoid, low quality) **bit **number of bits needed for the block **rd **rate distortion optimal, slow **zero** 0 **vsad** sum of absolute vertical differences **vsse** sum of squared vertical differences **nsse** noise preserving sum of squared differences **w53 **5/3 wavelet, only used in snow **w97 **9/7 wavelet, only used in snow **dctmax** **chroma** **pre_dia_size _**integer_ **(_**encoding,video_**)** Set diamond type & size for motion estimation pre-pass. **subq _**integer_ **(_**encoding,video_**)** Set sub pel motion estimation quality. **me_range _**integer_ **(_**encoding,video_**)** Set limit motion vectors range (1023 for DivX player). **global_quality _**integer_ **(_**encoding,audio,video_**)** **slice_flags _**integer_ **mbd _**integer_ **(_**encoding,video_**)** Set macroblock decision algorithm (high quality mode). Possible values: **simple** use mbcmp (default) **bits** use fewest bits **rd **use best rate distortion **rc_init_occupancy _**integer_ **(_**encoding,video_**)** Set number of bits which should be loaded into the rc buffer before decoding starts. **flags2 _**flags_ **(_**decoding/encoding,audio,video,subtitles_**)** Possible values: **fast** Allow non spec compliant speedup tricks. **noout** Skip bitstream encoding. **ignorecrop** Ignore cropping information from sps. **local_header** Place global headers at every keyframe instead of in extradata. **chunks** Frame data might be split into multiple chunks. **showall** Show all frames before the first keyframe. **export_mvs** Export motion vectors into frame side-data (see "AV_FRAME_DATA_MOTION_VECTORS") for codecs that support it. See also _doc/examples/export_mvs.c_. **skip_manual** Do not skip samples and export skip information as frame side data. **ass_ro_flush_noop** Do not reset ASS ReadOrder field on flush. **icc_profiles** Generate/parse embedded ICC profiles from/to colorimetry tags. **export_side_data _**flags_ **(_**decoding/encoding,audio,video,subtitles_**)** Possible values: **mvs **Export motion vectors into frame side-data (see "AV_FRAME_DATA_MOTION_VECTORS") for codecs that support it. See also _doc/examples/export_mvs.c_. **prft** Export encoder Producer Reference Time into packet side-data (see "AV_PKT_DATA_PRFT") for codecs that support it. **venc_params** Export video encoding parameters through frame side data (see "AV_FRAME_DATA_VIDEO_ENC_PARAMS") for codecs that support it. At present, those are H.264 and VP9. **film_grain** Export film grain parameters through frame side data (see "AV_FRAME_DATA_FILM_GRAIN_PARAMS"). Supported at present by AV1 decoders. **threads _**integer_ **(_**decoding/encoding,video_**)** Set the number of threads to be used, in case the selected codec implementation supports multi-threading. Possible values: **auto, 0** automatically select the number of threads to set Default value is **auto**. **dc _**integer_ **(_**encoding,video_**)** Set intra_dc_precision. **nssew _**integer_ **(_**encoding,video_**)** Set nsse weight. **skip_top _**integer_ **(_**decoding,video_**)** Set number of macroblock rows at the top which are skipped. **skip_bottom _**integer_ **(_**decoding,video_**)** Set number of macroblock rows at the bottom which are skipped. **profile _**integer_ **(_**encoding,audio,video_**)** Set encoder codec profile. Default value is **unknown**. Encoder specific profiles are documented in the relevant encoder documentation. **level _**integer_ **(_**encoding,audio,video_**)** Set the encoder level. This level depends on the specific codec, and might correspond to the profile level. It is set by default to **unknown**. Possible values: **unknown** **lowres _**integer_ **(_**decoding,audio,video_**)** Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions. **mblmin _**integer_ **(_**encoding,video_**)** Set min macroblock lagrange factor (VBR). **mblmax _**integer_ **(_**encoding,video_**)** Set max macroblock lagrange factor (VBR). **skip_loop_filter _**integer_ **(_**decoding,video_**)** **skip_idct _**integer_ **(_**decoding,video_**)** **skip_frame _**integer_ **(_**decoding,video_**)** Make decoder discard processing depending on the frame type selected by the option value. **skip_loop_filter **skips frame loop filtering, **skip_idct **skips frame IDCT/dequantization, **skip_frame **skips decoding. Possible values: **none** Discard no frame. **default** Discard useless frames like 0-sized frames. **noref** Discard all non-reference frames. **bidir** Discard all bidirectional frames. **nokey** Discard all frames excepts keyframes. **nointra** Discard all frames except I frames. **all **Discard all frames. Default value is **default**. **bidir_refine _**integer_ **(_**encoding,video_**)** Refine the two motion vectors used in bidirectional macroblocks. **keyint_min _**integer_ **(_**encoding,video_**)** Set minimum interval between IDR-frames. **refs _**integer_ **(_**encoding,video_**)** Set reference frames to consider for motion compensation. **trellis _**integer_ **(_**encoding,audio,video_**)** Set rate-distortion optimal quantization. **mv0_threshold _**integer_ **(_**encoding,video_**)** **compression_level _**integer_ **(_**encoding,audio,video_**)** **bits_per_raw_sample _**integer_ **channel_layout _**integer_ **(_**decoding/encoding,audio_**)** Possible values: **request_channel_layout _**integer_ **(_**decoding,audio_**)** Possible values: **rc_max_vbv_use _**float_ **(_**encoding,video_**)** **rc_min_vbv_use _**float_ **(_**encoding,video_**)** **color_primaries _**integer_ **(_**decoding/encoding,video_**)** Possible values: **bt709** BT.709 **bt470m** BT.470 M **bt470bg** BT.470 BG **smpte170m** SMPTE 170 M **smpte240m** SMPTE 240 M **film** Film **bt2020** BT.2020 **smpte428** **smpte428_1** SMPTE ST 428-1 **smpte431** SMPTE 431-2 **smpte432** SMPTE 432-1 **jedec-p22** JEDEC P22 **color_trc _**integer_ **(_**decoding/encoding,video_**)** Possible values: **bt709** BT.709 **gamma22** BT.470 M **gamma28** BT.470 BG **smpte170m** SMPTE 170 M **smpte240m** SMPTE 240 M **linear** Linear **log** **log100** Log **log_sqrt** **log316** Log square root **iec61966_2_4** **iec61966-2-4** IEC 61966-2-4 **bt1361** **bt1361e** BT.1361 **iec61966_2_1** **iec61966-2-1** IEC 61966-2-1 **bt2020_10** **bt2020_10bit** BT.2020 - 10 bit **bt2020_12** **bt2020_12bit** BT.2020 - 12 bit **smpte2084** SMPTE ST 2084 **smpte428** **smpte428_1** SMPTE ST 428-1 **arib-std-b67** ARIB STD-B67 **colorspace _**integer_ **(_**decoding/encoding,video_**)** Possible values: **rgb **RGB **bt709** BT.709 **fcc **FCC **bt470bg** BT.470 BG **smpte170m** SMPTE 170 M **smpte240m** SMPTE 240 M **ycocg** YCOCG **bt2020nc** **bt2020_ncl** BT.2020 NCL **bt2020c** **bt2020_cl** BT.2020 CL **smpte2085** SMPTE 2085 **chroma-derived-nc** Chroma-derived NCL **chroma-derived-c** Chroma-derived CL **ictcp** ICtCp **color_range _**integer_ **(_**decoding/encoding,video_**)** If used as input parameter, it serves as a hint to the decoder, which color_range the input has. Possible values: **tv** **mpeg** **limited** MPEG (219*2^(n-8)) **pc** **jpeg** **full** JPEG (2^n-1) **chroma_sample_location _**integer_ **(_**decoding/encoding,video_**)** Possible values: **left** **center** **topleft** **top** **bottomleft** **bottom** **log_level_offset _**integer_ Set the log level offset. **slices _**integer_ **(_**encoding,video_**)** Number of slices, used in parallelized encoding. **thread_type _**flags_ **(_**decoding/encoding,video_**)** Select which multithreading methods to use. Use of **frame **will increase decoding delay by one frame per thread, so clients which cannot provide future frames should not use it. Possible values: **slice** Decode more than one part of a single frame at once. Multithreading using slices works only when the video was encoded with slices. **frame** Decode more than one frame at once. Default value is **slice+frame**. **audio_service_type _**integer_ **(_**encoding,audio_**)** Set audio service type. Possible values: **ma **Main Audio Service **ef **Effects **vi **Visually Impaired **hi **Hearing Impaired **di **Dialogue **co **Commentary **em **Emergency **vo **Voice Over **ka **Karaoke **request_sample_fmt _**sample_fmt_ **(_**decoding,audio_**)** Set sample format audio decoders should prefer. Default value is "none". **pkt_timebase _**rational_ _number_ **sub_charenc _**encoding_ **(_**decoding,subtitles_**)** Set the input subtitles character encoding. **field_order _**field_order_ **(_**video_**)** Set/override the field order of the video. Possible values: **progressive** Progressive video **tt **Interlaced video, top field coded and displayed first **bb **Interlaced video, bottom field coded and displayed first **tb **Interlaced video, top coded first, bottom displayed first **bt **Interlaced video, bottom coded first, top displayed first **skip_alpha _**bool_ **(_**decoding,video_**)** Set to 1 to disable processing alpha (transparency). This works like the **gray **flag in the **flags **option which skips chroma information instead of alpha. Default is 0. **codec_whitelist _**list_ **(_**input_**)** "," separated list of allowed decoders. By default all are allowed. **dump_separator _**string_ **(_**input_**)** Separator used to separate the fields printed on the command line about the Stream parameters. For example, to separate the fields with newlines and indentation: ffprobe -dump_separator " " -i ~/videos/matrixbench_mpeg2.mpg **max_pixels _**integer_ **(_**decoding/encoding,video_**)** Maximum number of pixels per image. This value can be used to avoid out of memory failures due to large images. **apply_cropping _**bool_ **(_**decoding,video_**)** Enable cropping if cropping parameters are multiples of the required alignment for the left and top parameters. If the alignment is not met the cropping will be partially applied to maintain alignment. Default is 1 (enabled). Note: The required alignment depends on if "AV_CODEC_FLAG_UNALIGNED" is set and the CPU. "AV_CODEC_FLAG_UNALIGNED" cannot be changed from the command line. Also hardware decoders will not apply left/top Cropping. ## DECODERS Decoders are configured elements in FFmpeg which allow the decoding of multimedia streams. When you configure your FFmpeg build, all the supported native decoders are enabled by default. Decoders requiring an external library must be enabled manually via the corresponding "--enable-lib" option. You can list all available decoders using the configure option "--list-decoders". You can disable all the decoders with the configure option "--disable-decoders" and selectively enable / disable single decoders with the options "--enable-decoder=_DECODER_" / "--disable-decoder=_DECODER_". The option "-decoders" of the ff* tools will display the list of enabled decoders. ## VIDEO DECODERS A description of some of the currently available video decoders follows. ### av1 AOMedia Video 1 (AV1) decoder. _Options_ ### operating_point Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0. ### rawvideo Raw video decoder. This decoder decodes rawvideo streams. _Options_ **top _**top_field_first_ Specify the assumed field type of the input video. **-1 **the video is assumed to be progressive (default) **0 **bottom-field-first is assumed **1 **top-field-first is assumed ### libdav1d dav1d AV1 decoder. libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec. Requires the presence of the libdav1d headers and library during configuration. You need to explicitly configure the build with "--enable-libdav1d". _Options_ The following options are supported by the libdav1d wrapper. ### framethreads Set amount of frame threads to use during decoding. The default value is 0 (autodetect). This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the option "max_frame_delay" and the global option "threads" instead. ### tilethreads Set amount of tile threads to use during decoding. The default value is 0 (autodetect). This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the global option "threads" instead. ### max_frame_delay Set max amount of frames the decoder may buffer internally. The default value is 0 (autodetect). ### filmgrain Apply film grain to the decoded video if present in the bitstream. Defaults to the internal default of the library. This option is deprecated and will be removed in the future. See the global option "export_side_data" to export Film Grain parameters instead of applying it. ### oppoint Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the internal default of the library. ### alllayers Output all spatial layers of a scalable AV1 bitstream. The default value is false. ### libdavs2 AVS2-P2/IEEE1857.4 video decoder wrapper. This decoder allows libavcodec to decode AVS2 streams with davs2 library. ### libuavs3d AVS3-P2/IEEE1857.10 video decoder. libuavs3d allows libavcodec to decode AVS3 streams. Requires the presence of the libuavs3d headers and library during configuration. You need to explicitly configure the build with "--enable-libuavs3d". _Options_ The following option is supported by the libuavs3d wrapper. ### frame_threads Set amount of frame threads to use during decoding. The default value is 0 (autodetect). ### QSV Decoders The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC, JPEG/MJPEG, VP8, VP9, AV1). _Common_ _Options_ The following options are supported by all qsv decoders. _async_depth_ Internal parallelization depth, the higher the value the higher the latency. _gpu_copy_ A GPU-accelerated copy between video and system memory **default** **on** **off** _HEVC_ _Options_ Extra options for hevc_qsv. _load_plugin_ A user plugin to load in an internal session **none** **hevc_sw** **hevc_hw** _load_plugins_ A :-separate list of hexadecimal plugin UIDs to load in an internal session ### v210 Uncompressed 4:2:2 10-bit decoder. _Options_ ### custom_stride Set the line size of the v210 data in bytes. The default value is 0 (autodetect). You can use the special -1 value for a strideless v210 as seen in BOXX files. ## AUDIO DECODERS A description of some of the currently available audio decoders follows. ### ac3 AC-3 audio decoder. This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet). _AC-3_ _Decoder_ _Options_ ### -drc_scale _ Dynamic Range Scale Factor. The factor to apply to dynamic range values from the AC-3 stream. This factor is applied exponentially. The default value is 1. There are 3 notable scale factor ranges: **drc_scale == 0** DRC disabled. Produces full range audio. **0 < drc_scale <= 1** DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full range and full compression. **drc_scale > 1** DRC enabled. Applies drc_scale asymmetrically. Loud sounds are fully compressed. Soft sounds are enhanced. ### flac FLAC audio decoder. This decoder aims to implement the complete FLAC specification from Xiph. _FLAC_ _Decoder_ _options_ ### -use_buggy_lpc The lavc FLAC encoder used to produce buggy streams with high lpc values (like the default value). This option makes it possible to decode such streams correctly by using lavc's old buggy lpc logic for decoding. ### ffwavesynth Internal wave synthesizer. This decoder generates wave patterns according to predefined sequences. Its use is purely internal and the format of the data it accepts is not publicly documented. ### libcelt libcelt decoder wrapper. libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec. Requires the presence of the libcelt headers and library during configuration. You need to explicitly configure the build with "--enable-libcelt". ### libgsm libgsm decoder wrapper. libgsm allows libavcodec to decode the GSM full rate audio codec. Requires the presence of the libgsm headers and library during configuration. You need to explicitly configure the build with "--enable-libgsm". This decoder supports both the ordinary GSM and the Microsoft variant. ### libilbc libilbc decoder wrapper. libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC) audio codec. Requires the presence of the libilbc headers and library during configuration. You need to explicitly configure the build with "--enable-libilbc". _Options_ The following option is supported by the libilbc wrapper. ### enhance Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled). ### libopencore-amrnb libopencore-amrnb decoder wrapper. libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate Narrowband audio codec. Using it requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with "--enable-libopencore-amrnb". An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB without this library. ### libopencore-amrwb libopencore-amrwb decoder wrapper. libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate Wideband audio codec. Using it requires the presence of the libopencore-amrwb headers and library during configuration. You need to explicitly configure the build with "--enable-libopencore-amrwb". An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB without this library. ### libopus libopus decoder wrapper. libopus allows libavcodec to decode the Opus Interactive Audio Codec. Requires the presence of the libopus headers and library during configuration. You need to explicitly configure the build with "--enable-libopus". An FFmpeg native decoder for Opus exists, so users can decode Opus without this library. ## SUBTITLES DECODERS ### libaribb24 ARIB STD-B24 caption decoder. Implements profiles A and C of the ARIB STD-B24 standard. _libaribb24_ _Decoder_ _Options_ ### -aribb24-base-path _ Sets the base path for the libaribb24 library. This is utilized for reading of configuration files (for custom unicode conversions), and for dumping of non-text symbols as images under that location. Unset by default. ### -aribb24-skip-ruby-text _ Tells the decoder wrapper to skip text blocks that contain half-height ruby text. Enabled by default. ### libaribcaption Yet another ARIB STD-B24 caption decoder using external _libaribcaption_ library. Implements profiles A and C of the Japanse ARIB STD-B24 standard, Brazilian ABNT NBR 15606-1, and Philippines version of ISDB-T. Requires the presence of the libaribcaption headers and library (<****>) during configuration. You need to explicitly configure the build with "--enable-libaribcaption". If both _libaribb24_ and _libaribcaption_ are enabled, _libaribcaption_ decoder precedes. _libaribcaption_ _Decoder_ _Options_ ### -sub_type _ Specifies the format of the decoded subtitles. **bitmap** Graphical image. **ass **ASS formatted text. **text** Simple text based output without formatting. The default is _ass_ as same as _libaribb24_ decoder. Some present players (e.g., _mpv_) expect ASS format for ARIB caption. ### -caption_encoding _ Specifies the encoding scheme of input subtitle text. **auto** Automatically detect text encoding (default). **jis **8bit-char JIS encoding defined in ARIB STD B24. This encoding used in Japan for ISDB captions. **utf8** UTF-8 encoding defined in ARIB STD B24. This encoding is used in Philippines for ISDB-T captions. **latin** Latin character encoding defined in ABNT NBR 15606-1. This encoding is used in South America for SBTVD / ISDB-Tb captions. ### -font _ Specify comma-separated list of font family names to be used for _bitmap_ or _ass_ type subtitle rendering. Only first font name is used for _ass_ type subtitle. If not specified, use internaly defined default font family. ### -ass_single_rect _ ARIB STD-B24 specifies that some captions may be displayed at different positions at a time (multi-rectangle subtitle). Since some players (e.g., old _mpv_) can't handle multiple ASS rectangles in a single AVSubtitle, or multiple ASS rectangles of indeterminate duration with the same start timestamp, this option can change the behavior so that all the texts are displayed in a single ASS rectangle. The default is _false_. If your player cannot handle AVSubtitles with multiple ASS rectangles properly, set this option to _true_ or define **ASS_SINGLE_RECT=1 **to change default behavior at compilation. ### -force_outline_text _ Specify whether always render outline text for all characters regardless of the indication by charactor style. The default is _false_. ### -outline_width _ Specify width for outline text, in dots (relative). The default is _1.5_. ### -ignore_background _ Specify whether to ignore background color rendering. The default is _false_. ### -ignore_ruby _ Specify whether to ignore rendering for ruby-like (furigana) characters. The default is _false_. ### -replace_drcs _ Specify whether to render replaced DRCS characters as Unicode characters. The default is _true_. ### -replace_msz_ascii _ Specify whether to replace MSZ (Middle Size; half width) fullwidth alphanumerics with halfwidth alphanumerics. The default is _true_. ### -replace_msz_japanese _ Specify whether to replace some MSZ (Middle Size; half width) fullwidth japanese special characters with halfwidth ones. The default is _true_. ### -replace_msz_glyph _ Specify whether to replace MSZ (Middle Size; half width) characters with halfwidth glyphs if the fonts supports it. This option works under FreeType or DirectWrite renderer with Adobe-Japan1 compliant fonts. e.g., IBM Plex Sans JP, Morisawa BIZ UDGothic, Morisawa BIZ UDMincho, Yu Gothic, Yu Mincho, and Meiryo. The default is _true_. ### -canvas_size _ Specify the resolution of the canvas to render subtitles to; usually, this should be frame size of input video. This only applies when "-subtitle_type" is set to _bitmap_. The libaribcaption decoder assumes input frame size for bitmap rendering as below: 1. PROFILE_A : 1440 x 1080 with SAR (PAR) 4:3 2. PROFILE_C : 320 x 180 with SAR (PAR) 1:1 If actual frame size of input video does not match above assumption, the rendered captions may be distorted. To make the captions undistorted, add "-canvas_size" option to specify actual input video size. Note that the "-canvas_size" option is not required for video with different size but same aspect ratio. In such cases, the caption will be stretched or shrunk to actual video size if "-canvas_size" option is not specified. If "-canvas_size" option is specified with different size, the caption will be stretched or shrunk as specified size with calculated SAR. _libaribcaption_ _decoder_ _usage_ _examples_ Display MPEG-TS file with ARIB subtitle by "ffplay" tool: ffplay -sub_type bitmap MPEG.TS Display MPEG-TS file with input frame size 1920x1080 by "ffplay" tool: ffplay -sub_type bitmap -canvas_size 1920x1080 MPEG.TS Embed ARIB subtitle in transcoded video: ffmpeg -sub_type bitmap -i src.m2t -filter_complex "[0:v][0:s]overlay" -vcodec h264 dest.mp4 ### dvbsub _Options_ ### compute_clut **-2 **Compute clut once if no matching CLUT is in the stream. **-1 **Compute clut if no matching CLUT is in the stream. **0 **Never compute CLUT **1 **Always compute CLUT and override the one provided in the stream. ### dvb_substream Selects the dvb substream, or all substreams if -1 which is default. ### dvdsub This codec decodes the bitmap subtitles used in DVDs; the same subtitles can also be found in VobSub file pairs and in some Matroska files. _Options_ ### palette Specify the global palette used by the bitmaps. When stored in VobSub, the palette is normally specified in the index file; in Matroska, the palette is stored in the codec extra-data in the same format as in VobSub. In DVDs, the palette is stored in the IFO file, and therefore not available when reading from dumped VOB files. The format for this option is a string containing 16 24-bits hexadecimal numbers (without 0x prefix) separated by commas, for example "0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b". ### ifo_palette Specify the IFO file from which the global palette is obtained. (experimental) ### forced_subs_only Only decode subtitle entries marked as forced. Some titles have forced and non-forced subtitles in the same track. Setting this flag to 1 will only keep the forced subtitles. Default value is 0. ### libzvbi-teletext Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext subtitles. Requires the presence of the libzvbi headers and library during configuration. You need to explicitly configure the build with "--enable-libzvbi". _Options_ ### txt_page List of teletext page numbers to decode. Pages that do not match the specified list are dropped. You may use the special "*" string to match all pages, or "subtitle" to match all subtitle pages. Default value is *. ### txt_default_region Set default character set used for decoding, a value between 0 and 87 (see ETS 300 706, Section 15, Table 32). Default value is -1, which does not override the libzvbi default. This option is needed for some legacy level 1.0 transmissions which cannot signal the proper charset. ### txt_chop_top Discards the top teletext line. Default value is 1. ### txt_format Specifies the format of the decoded subtitles. **bitmap** The default format, you should use this for teletext pages, because certain graphics and colors cannot be expressed in simple text or even ASS. **text** Simple text based output without formatting. **ass **Formatted ASS output, subtitle pages and teletext pages are returned in different styles, subtitle pages are stripped down to text, but an effort is made to keep the text alignment and the formatting. ### txt_left X offset of generated bitmaps, default is 0. ### txt_top Y offset of generated bitmaps, default is 0. ### txt_chop_spaces Chops leading and trailing spaces and removes empty lines from the generated text. This option is useful for teletext based subtitles where empty spaces may be present at the start or at the end of the lines or empty lines may be present between the subtitle lines because of double-sized teletext characters. Default value is 1. ### txt_duration Sets the display duration of the decoded teletext pages or subtitles in milliseconds. Default value is -1 which means infinity or until the next subtitle event comes. ### txt_transparent Force transparent background of the generated teletext bitmaps. Default value is 0 which means an opaque background. ### txt_opacity Sets the opacity (0-255) of the teletext background. If **txt_transparent **is not set, it only affects characters between a start box and an end box, typically subtitles. Default value is 0 if **txt_transparent **is set, 255 otherwise. ## ENCODERS Encoders are configured elements in FFmpeg which allow the encoding of multimedia streams. When you configure your FFmpeg build, all the supported native encoders are enabled by default. Encoders requiring an external library must be enabled manually via the corresponding "--enable-lib" option. You can list all available encoders using the configure option "--list-encoders". You can disable all the encoders with the configure option "--disable-encoders" and selectively enable / disable single encoders with the options "--enable-encoder=_ENCODER_" / "--disable-encoder=_ENCODER_". The option "-encoders" of the ff* tools will display the list of enabled encoders. ## AUDIO ENCODERS A description of some of the currently available audio encoders follows. ### aac Advanced Audio Coding (AAC) encoder. This encoder is the default AAC encoder, natively implemented into FFmpeg. _Options_ **b **Set bit rate in bits/s. Setting this automatically activates constant bit rate (CBR) mode. If this option is unspecified it is set to 128kbps. **q **Set quality for variable bit rate (VBR) mode. This option is valid only using the **ffmpeg** command-line tool. For library interface users, use **global_quality**. ### cutoff Set cutoff frequency. If unspecified will allow the encoder to dynamically adjust the cutoff to improve clarity on low bitrates. ### aac_coder Set AAC encoder coding method. Possible values: **twoloop** Two loop searching (TLS) method. This is the default method. This method first sets quantizers depending on band thresholds and then tries to find an optimal combination by adding or subtracting a specific value from all quantizers and adjusting some individual quantizer a little. Will tune itself based on whether **aac_is**, **aac_ms **and **aac_pns **are enabled. **anmr** Average noise to mask ratio (ANMR) trellis-based solution. This is an experimental coder which currently produces a lower quality, is more unstable and is slower than the default twoloop coder but has potential. Currently has no support for the **aac_is **or **aac_pns **options. Not currently recommended. **fast** Constant quantizer method. Uses a cheaper version of twoloop algorithm that doesn't try to do as many clever adjustments. Worse with low bitrates (less than 64kbps), but is better and much faster at higher bitrates. ### aac_ms Sets mid/side coding mode. The default value of "auto" will automatically use M/S with bands which will benefit from such coding. Can be forced for all bands using the value "enable", which is mainly useful for debugging or disabled using "disable". ### aac_is Sets intensity stereo coding tool usage. By default, it's enabled and will automatically toggle IS for similar pairs of stereo bands if it's beneficial. Can be disabled for debugging by setting the value to "disable". ### aac_pns Uses perceptual noise substitution to replace low entropy high frequency bands with imperceptible white noise during the decoding process. By default, it's enabled, but can be disabled for debugging purposes by using "disable". ### aac_tns Enables the use of a multitap FIR filter which spans through the high frequency bands to hide quantization noise during the encoding process and is reverted by the decoder. As well as decreasing unpleasant artifacts in the high range this also reduces the entropy in the high bands and allows for more bits to be used by the mid-low bands. By default it's enabled but can be disabled for debugging by setting the option to "disable". ### aac_ltp Enables the use of the long term prediction extension which increases coding efficiency in very low bandwidth situations such as encoding of voice or solo piano music by extending constant harmonic peaks in bands throughout frames. This option is implied by profile:a aac_low and is incompatible with aac_pred. Use in conjunction with **-ar **to decrease the samplerate. ### aac_pred Enables the use of a more traditional style of prediction where the spectral coefficients transmitted are replaced by the difference of the current coefficients minus the previous "predicted" coefficients. In theory and sometimes in practice this can improve quality for low to mid bitrate audio. This option implies the aac_main profile and is incompatible with aac_ltp. ### profile Sets the encoding profile, possible values: **aac_low** The default, AAC "Low-complexity" profile. Is the most compatible and produces decent quality. **mpeg2_aac_low** Equivalent to "-profile:a aac_low -aac_pns 0". PNS was introduced with the MPEG4 specifications. **aac_ltp** Long term prediction profile, is enabled by and will enable the **aac_ltp **option. Introduced in MPEG4. **aac_main** Main-type prediction profile, is enabled by and will enable the **aac_pred **option. Introduced in MPEG2. If this option is unspecified it is set to **aac_low**. ### ac3 and ac3_fixed AC-3 audio encoders. These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet). The _ac3_ encoder uses floating-point math, while the _ac3_fixed_ encoder only uses fixed-point integer math. This does not mean that one is always faster, just that one or the other may be better suited to a particular system. The _ac3_fixed_ encoder is not the default codec for any of the output formats, so it must be specified explicitly using the option "-acodec ac3_fixed" in order to use it. _AC-3_ _Metadata_ The AC-3 metadata options are used to set parameters that describe the audio, but in most cases do not affect the audio encoding itself. Some of the options do directly affect or influence the decoding and playback of the resulting bitstream, while others are just for informational purposes. A few of the options will add bits to the output stream that could otherwise be used for audio data, and will thus affect the quality of the output. Those will be indicated accordingly with a note in the option list below. These parameters are described in detail in several publicly-available documents. *<<****>> *<<****>> *<<****>> *<<****>> Metadata Control Options ### -per_frame_metadata _ Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame. **0 **The metadata values set at initialization will be used for every frame in the stream. (default) **1 **Metadata values can be changed before encoding each frame. Downmix Levels ### -center_mixlev _ Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo. This field will only be written to the bitstream if a center channel is present. The value is specified as a scale factor. There are 3 valid values: **0.707** Apply -3dB gain **0.595** Apply -4.5dB gain (default) **0.500** Apply -6dB gain ### -surround_mixlev _ Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo. This field will only be written to the bitstream if one or more surround channels are present. The value is specified as a scale factor. There are 3 valid values: **0.707** Apply -3dB gain **0.500** Apply -6dB gain (default) **0.000** Silence Surround Channel(s) Audio Production Information Audio Production Information is optional information describing the mixing environment. Either none or both of the fields are written to the bitstream. ### -mixing_level _ Mixing Level. Specifies peak sound pressure level (SPL) in the production environment when the mix was mastered. Valid values are 80 to 111, or -1 for unknown or not indicated. The default value is -1, but that value cannot be used if the Audio Production Information is written to the bitstream. Therefore, if the "room_type" option is not the default value, the "mixing_level" option must not be -1. ### -room_type _ Room Type. Describes the equalization used during the final mixing session at the studio or on the dubbing stage. A large room is a dubbing stage with the industry standard X-curve equalization; a small room has flat equalization. This field will not be written to the bitstream if both the "mixing_level" option and the "room_type" option have the default values. **0** **notindicated** Not Indicated (default) **1** **large** Large Room **2** **small** Small Room Other Metadata Options ### -copyright _ Copyright Indicator. Specifies whether a copyright exists for this audio. **0** **off **No Copyright Exists (default) **1** **on **Copyright Exists ### -dialnorm _ Dialogue Normalization. Indicates how far the average dialogue level of the program is below digital 100% full scale (0 dBFS). This parameter determines a level shift during audio reproduction that sets the average volume of the dialogue to a preset level. The goal is to match volume level between program sources. A value of -31dB will result in no volume level change, relative to the source volume, during audio reproduction. Valid values are whole numbers in the range -31 to -1, with -31 being the default. ### -dsur_mode _ Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround (Pro Logic). This field will only be written to the bitstream if the audio stream is stereo. Using this option does **NOT **mean the encoder will actually apply Dolby Surround processing. **0** **notindicated** Not Indicated (default) **1** **off **Not Dolby Surround Encoded **2** **on **Dolby Surround Encoded ### -original _ Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy. **0** **off **Not Original Source **1** **on **Original Source (default) _Extended_ _Bitstream_ _Information_ The extended bitstream options are part of the Alternate Bit Stream Syntax as specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts. If any one parameter in a group is specified, all values in that group will be written to the bitstream. Default values are used for those that are written but have not been specified. If the mixing levels are written, the decoder will use these values instead of the ones specified in the "center_mixlev" and "surround_mixlev" options if it supports the Alternate Bit Stream Syntax. Extended Bitstream Information - Part 1 ### -dmix_mode _ Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt (Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode. **0** **notindicated** Not Indicated (default) **1** **ltrt** Lt/Rt Downmix Preferred **2** **loro** Lo/Ro Downmix Preferred ### -ltrt_cmixlev _ Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lt/Rt mode. **1.414** Apply +3dB gain **1.189** Apply +1.5dB gain **1.000** Apply 0dB gain **0.841** Apply -1.5dB gain **0.707** Apply -3.0dB gain **0.595** Apply -4.5dB gain (default) **0.500** Apply -6.0dB gain **0.000** Silence Center Channel ### -ltrt_surmixlev _ Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lt/Rt mode. **0.841** Apply -1.5dB gain **0.707** Apply -3.0dB gain **0.595** Apply -4.5dB gain **0.500** Apply -6.0dB gain (default) **0.000** Silence Surround Channel(s) ### -loro_cmixlev _ Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lo/Ro mode. **1.414** Apply +3dB gain **1.189** Apply +1.5dB gain **1.000** Apply 0dB gain **0.841** Apply -1.5dB gain **0.707** Apply -3.0dB gain **0.595** Apply -4.5dB gain (default) **0.500** Apply -6.0dB gain **0.000** Silence Center Channel ### -loro_surmixlev _ Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lo/Ro mode. **0.841** Apply -1.5dB gain **0.707** Apply -3.0dB gain **0.595** Apply -4.5dB gain **0.500** Apply -6.0dB gain (default) **0.000** Silence Surround Channel(s) Extended Bitstream Information - Part 2 ### -dsurex_mode _ Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX (7.1 matrixed to 5.1). Using this option does **NOT **mean the encoder will actually apply Dolby Surround EX processing. **0** **notindicated** Not Indicated (default) **1** **on **Dolby Surround EX Off **2** **off **Dolby Surround EX On ### -dheadphone_mode _ Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone encoding (multi- channel matrixed to 2.0 for use with headphones). Using this option does **NOT **mean the encoder will actually apply Dolby Headphone processing. **0** **notindicated** Not Indicated (default) **1** **on **Dolby Headphone Off **2** **off **Dolby Headphone On ### -ad_conv_type _ A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion. **0** **standard** Standard A/D Converter (default) **1** **hdcd** HDCD A/D Converter _Other_ _AC-3_ _Encoding_ _Options_ ### -stereo_rematrixing _ Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This is an optional AC-3 feature that increases quality by selectively encoding the left/right channels as mid/side. This option is enabled by default, and it is highly recommended that it be left as enabled except for testing purposes. **cutoff _**frequency_ Set lowpass cutoff frequency. If unspecified, the encoder selects a default determined by various other encoding parameters. _Floating-Point-Only_ _AC-3_ _Encoding_ _Options_ These options are only valid for the floating-point encoder and do not exist for the fixed- point encoder due to the corresponding features not being implemented in fixed-point. ### -channel_coupling _ Enables/Disables use of channel coupling, which is an optional AC-3 feature that increases quality by combining high frequency information from multiple channels into a single channel. The per-channel high frequency information is sent with less accuracy in both the frequency and time domains. This allows more bits to be used for lower frequencies while preserving enough information to reconstruct the high frequencies. This option is enabled by default for the floating-point encoder and should generally be left as enabled except for testing purposes or to increase encoding speed. **-1** **auto** Selected by Encoder (default) **0** **off **Disable Channel Coupling **1** **on **Enable Channel Coupling ### -cpl_start_band _ Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a value higher than the bandwidth is used, it will be reduced to 1 less than the coupling end band. If _auto_ is used, the start band will be determined by the encoder based on the bit rate, sample rate, and channel layout. This option has no effect if channel coupling is disabled. **-1** **auto** Selected by Encoder (default) ### flac FLAC (Free Lossless Audio Codec) Encoder _Options_ The following options are supported by FFmpeg's flac encoder. ### compression_level Sets the compression level, which chooses defaults for many other options if they are not set explicitly. Valid values are from 0 to 12, 5 is the default. ### frame_size Sets the size of the frames in samples per channel. ### lpc_coeff_precision Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the default. ### lpc_type Sets the first stage LPC algorithm **none** LPC is not used **fixed** fixed LPC coefficients **levinson** **cholesky** ### lpc_passes Number of passes to use for Cholesky factorization during LPC analysis ### min_partition_order The minimum partition order ### max_partition_order The maximum partition order ### prediction_order_method **estimation** **2level** **4level** **8level** **search** Bruteforce search **log** ### ch_mode Channel mode **auto** The mode is chosen automatically for each frame **indep** Channels are independently coded **left_side** **right_side** **mid_side** ### exact_rice_parameters Chooses if rice parameters are calculated exactly or approximately. if set to 1 then they are chosen exactly, which slows the code down slightly and improves compression slightly. ### multi_dim_quant Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is applied after the first stage to finetune the coefficients. This is quite slow and slightly improves compression. ### opus Opus encoder. This is a native FFmpeg encoder for the Opus format. Currently its in development and only implements the CELT part of the codec. Its quality is usually worse and at best is equal to the libopus encoder. _Options_ **b **Set bit rate in bits/s. If unspecified it uses the number of channels and the layout to make a good guess. ### opus_delay Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly decrease quality. ### libfdk_aac libfdk-aac AAC (Advanced Audio Coding) encoder wrapper. The libfdk-aac library is based on the Fraunhofer FDK AAC code from the Android project. Requires the presence of the libfdk-aac headers and library during configuration. You need to explicitly configure the build with "--enable-libfdk-aac". The library is also incompatible with GPL, so if you allow the use of GPL, you should configure with "--enable-gpl --enable-nonfree --enable-libfdk-aac". This encoder has support for the AAC-HE profiles. VBR encoding, enabled through the **vbr **or **flags +qscale **options, is experimental and only works with some combinations of parameters. Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or higher. For more information see the fdk-aac project at <****>. _Options_ The following options are mapped on the shared FFmpeg codec options. **b **Set bit rate in bits/s. If the bitrate is not explicitly specified, it is automatically set to a suitable value depending on the selected profile. In case VBR mode is enabled the option is ignored. **ar **Set audio sampling rate (in Hz). ### channels Set the number of audio channels. ### flags +qscale Enable fixed quality, VBR (Variable Bit Rate) mode. Note that VBR is implicitly enabled when the **vbr **value is positive. ### cutoff Set cutoff frequency. If not specified (or explicitly set to 0) it will use a value automatically computed by the library. Default value is 0. ### profile Set audio profile. The following profiles are recognized: **aac_low** Low Complexity AAC (LC) **aac_he** High Efficiency AAC (HE-AAC) **aac_he_v2** High Efficiency AAC version 2 (HE-AACv2) **aac_ld** Low Delay AAC (LD) **aac_eld** Enhanced Low Delay AAC (ELD) If not specified it is set to **aac_low**. The following are private options of the libfdk_aac encoder. ### afterburner Enable afterburner feature if set to 1, disabled if set to 0. This improves the quality but also the required processing power. Default value is 1. ### eld_sbr Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled if set to 0. Default value is 0. ### eld_v2 Enable ELDv2 (LD-MPS extension for ELD stereo signals) for ELDv2 if set to 1, disabled if set to 0. Note that option is available when fdk-aac version (AACENCODER_LIB_VL0.AACENCODER_LIB_VL1.AACENCODER_LIB_VL2) > (4.0.0). Default value is 0. ### signaling Set SBR/PS signaling style. It can assume one of the following values: **default** choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled) **implicit** implicit backwards compatible signaling **explicit_sbr** explicit SBR, implicit PS signaling **explicit_hierarchical** explicit hierarchical signaling Default value is **default**. ### latm Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0. Default value is 0. ### header_period Set StreamMuxConfig and PCE repetition period (in frames) for sending in-band configuration buffers within LATM/LOAS transport layer. Must be a 16-bits non-negative integer. Default value is 0. **vbr **Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty good) and 5 is highest quality. A value of 0 will disable VBR, and CBR (Constant Bit Rate) is enabled. Currently only the **aac_low **profile supports VBR encoding. VBR modes 1-5 correspond to roughly the following average bit rates: **1 **32 kbps/channel **2 **40 kbps/channel **3 **48-56 kbps/channel **4 **64 kbps/channel **5 **about 80-96 kbps/channel Default value is 0. ### frame_length Set the audio frame length in samples. Default value is the internal default of the library. Refer to the library's documentation for information about supported values. _Examples_ • Use **ffmpeg **to convert an audio file to VBR AAC in an M4A (MP4) container: ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a • Use **ffmpeg **to convert an audio file to CBR 64k kbps AAC, using the High-Efficiency AAC profile: ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a ### libmp3lame LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper. Requires the presence of the libmp3lame headers and library during configuration. You need to explicitly configure the build with "--enable-libmp3lame". See **libshine **for a fixed-point MP3 encoder, although with a lower quality. _Options_ The following options are supported by the libmp3lame wrapper. The **lame**-equivalent of the options are listed in parentheses. **b (_**-b_**)** Set bitrate expressed in bits/s for CBR or ABR. LAME "bitrate" is expressed in kilobits/s. **q (_**-V_**)** Set constant quality setting for VBR. This option is valid only using the **ffmpeg **command- line tool. For library interface users, use **global_quality**. **compression_level (_**-q_**)** Set algorithm quality. Valid arguments are integers in the 0-9 range, with 0 meaning highest quality but slowest, and 9 meaning fastest while producing the worst quality. **cutoff (_**--lowpass_**)** Set lowpass cutoff frequency. If unspecified, the encoder dynamically adjusts the cutoff. ### reservoir Enable use of bit reservoir when set to 1. Default value is 1. LAME has this enabled by default, but can be overridden by use **--nores **option. **joint_stereo (_**-m_ _j_**)** Enable the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo. Default value is 1. **abr (_**--abr_**)** Enable the encoder to use ABR when set to 1. The **lame --abr **sets the target bitrate, while this options only tells FFmpeg to use ABR still relies on **b **to set bitrate. **copyright (_**-c_**)** Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled). **original (_**-o_**)** Set MPEG audio original flag when set to 1. The default value is 1 (enabled). ### libopencore-amrnb OpenCORE Adaptive Multi-Rate Narrowband encoder. Requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with "--enable-libopencore-amrnb --enable-version3". This is a mono-only encoder. Officially it only supports 8000Hz sample rate, but you can override it by setting **strict **to **unofficial **or lower. _Options_ **b **Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate. **4750** **5150** **5900** **6700** **7400** **7950** **10200** **12200** **dtx **Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled). ### libopus libopus Opus Interactive Audio Codec encoder wrapper. Requires the presence of the libopus headers and library during configuration. You need to explicitly configure the build with "--enable-libopus". _Option_ _Mapping_ Most libopus options are modelled after the **opusenc **utility from opus-tools. The following is an option mapping chart describing options supported by the libopus wrapper, and their **opusenc**-equivalent in parentheses. **b (_**bitrate_**)** Set the bit rate in bits/s. FFmpeg's **b **option is expressed in bits/s, while **opusenc**'s **bitrate **in kilobits/s. **vbr (_**vbr_**, _**hard-cbr_**, and _**cvbr_**)** Set VBR mode. The FFmpeg **vbr **option has the following valid arguments, with the **opusenc** equivalent options in parentheses: **off (_**hard-cbr_**)** Use constant bit rate encoding. **on (_**vbr_**)** Use variable bit rate encoding (the default). **constrained (_**cvbr_**)** Use constrained variable bit rate encoding. **compression_level (_**comp_**)** Set encoding algorithm complexity. Valid options are integers in the 0-10 range. 0 gives the fastest encodes but lower quality, while 10 gives the highest quality but slowest encoding. The default is 10. **frame_duration (_**framesize_**)** Set maximum frame size, or duration of a frame in milliseconds. The argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller frame sizes achieve lower latency but less quality at a given bitrate. Sizes greater than 20ms are only interesting at fairly low bitrates. The default is 20ms. **packet_loss (_**expect-loss_**)** Set expected packet loss percentage. The default is 0. **fec (_**n/a_**)** Enable inband forward error correction. **packet_loss **must be non-zero to take advantage - frequency of FEC 'side-data' is proportional to expected packet loss. Default is disabled. ### application (N.A.) Set intended application type. Valid options are listed below: **voip** Favor improved speech intelligibility. **audio** Favor faithfulness to the input (the default). **lowdelay** Restrict to only the lowest delay modes by disabling voice-optimized modes. ### cutoff (N.A.) Set cutoff bandwidth in Hz. The argument must be exactly one of the following: 4000, 6000, 8000, 12000, or 20000, corresponding to narrowband, mediumband, wideband, super wideband, and fullband respectively. The default is 0 (cutoff disabled). Note that libopus forces a wideband cutoff for bitrates < 15 kbps, unless CELT-only (**application** set to **lowdelay**) mode is used. **mapping_family (_**mapping_family_**)** Set channel mapping family to be used by the encoder. The default value of -1 uses mapping family 0 for mono and stereo inputs, and mapping family 1 otherwise. The default also disables the surround masking and LFE bandwidth optimzations in libopus, and requires that the input contains 8 channels or fewer. Other values include 0 for mono and stereo, 1 for surround sound with masking and LFE bandwidth optimizations, and 255 for independent streams with an unspecified channel layout. ### apply_phase_inv (N.A.) (requires libopus >= 1.2) If set to 0, disables the use of phase inversion for intensity stereo, improving the quality of mono downmixes, but slightly reducing normal stereo quality. The default is 1 (phase inversion enabled). ### libshine Shine Fixed-Point MP3 encoder wrapper. Shine is a fixed-point MP3 encoder. It has a far better performance on platforms without an FPU, e.g. armel CPUs, and some phones and tablets. However, as it is more targeted on performance than quality, it is not on par with LAME and other production-grade encoders quality-wise. Also, according to the project's homepage, this encoder may not be free of bugs as the code was written a long time ago and the project was dead for at least 5 years. This encoder only supports stereo and mono input. This is also CBR-only. The original project (last updated in early 2007) is at <****>. We only support the updated fork by the Savonet/Liquidsoap project at <****>. Requires the presence of the libshine headers and library during configuration. You need to explicitly configure the build with "--enable-libshine". See also **libmp3lame**. _Options_ The following options are supported by the libshine wrapper. The **shineenc**-equivalent of the options are listed in parentheses. **b (_**-b_**)** Set bitrate expressed in bits/s for CBR. **shineenc -b **option is expressed in kilobits/s. ### libtwolame TwoLAME MP2 encoder wrapper. Requires the presence of the libtwolame headers and library during configuration. You need to explicitly configure the build with "--enable-libtwolame". _Options_ The following options are supported by the libtwolame wrapper. The **twolame**-equivalent options follow the FFmpeg ones and are in parentheses. **b (_**-b_**)** Set bitrate expressed in bits/s for CBR. **twolame b **option is expressed in kilobits/s. Default value is 128k. **q (_**-V_**)** Set quality for experimental VBR support. Maximum value range is from -50 to 50, useful range is from -10 to 10. The higher the value, the better the quality. This option is valid only using the **ffmpeg **command-line tool. For library interface users, use **global_quality**. **mode (_**--mode_**)** Set the mode of the resulting audio. Possible values: **auto** Choose mode automatically based on the input. This is the default. **stereo** Stereo **joint_stereo** Joint stereo **dual_channel** Dual channel **mono** Mono **psymodel (_**--psyc-mode_**)** Set psychoacoustic model to use in encoding. The argument must be an integer between -1 and 4, inclusive. The higher the value, the better the quality. The default value is 3. **energy_levels (_**--energy_**)** Enable energy levels extensions when set to 1. The default value is 0 (disabled). **error_protection (_**--protect_**)** Enable CRC error protection when set to 1. The default value is 0 (disabled). **copyright (_**--copyright_**)** Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled). **original (_**--original_**)** Set MPEG audio original flag when set to 1. The default value is 0 (disabled). ### libvo-amrwbenc VisualOn Adaptive Multi-Rate Wideband encoder. Requires the presence of the libvo-amrwbenc headers and library during configuration. You need to explicitly configure the build with "--enable-libvo-amrwbenc --enable-version3". This is a mono-only encoder. Officially it only supports 16000Hz sample rate, but you can override it by setting **strict **to **unofficial **or lower. _Options_ **b **Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate. **6600** **8850** **12650** **14250** **15850** **18250** **19850** **23050** **23850** **dtx **Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled). ### libvorbis libvorbis encoder wrapper. Requires the presence of the libvorbisenc headers and library during configuration. You need to explicitly configure the build with "--enable-libvorbis". _Options_ The following options are supported by the libvorbis wrapper. The **oggenc**-equivalent of the options are listed in parentheses. To get a more accurate and extensive documentation of the libvorbis options, consult the libvorbisenc's and **oggenc**'s documentations. See <****>, <****>, and [**oggenc**(1)](https://www.chedong.com/phpMan.php/man/oggenc/1/markdown). **b (_**-b_**)** Set bitrate expressed in bits/s for ABR. **oggenc -b **is expressed in kilobits/s. **q (_**-q_**)** Set constant quality setting for VBR. The value should be a float number in the range of -1.0 to 10.0. The higher the value, the better the quality. The default value is **3.0**. This option is valid only using the **ffmpeg **command-line tool. For library interface users, use **global_quality**. **cutoff (_**--advanced-encode-option_ _lowpass_frequency=N_**)** Set cutoff bandwidth in Hz, a value of 0 disables cutoff. **oggenc**'s related option is expressed in kHz. The default value is **0 **(cutoff disabled). **minrate (_**-m_**)** Set minimum bitrate expressed in bits/s. **oggenc -m **is expressed in kilobits/s. **maxrate (_**-M_**)** Set maximum bitrate expressed in bits/s. **oggenc -M **is expressed in kilobits/s. This only has effect on ABR mode. **iblock (_**--advanced-encode-option_ _impulse_noisetune=N_**)** Set noise floor bias for impulse blocks. The value is a float number from -15.0 to 0.0. A negative bias instructs the encoder to pay special attention to the crispness of transients in the encoded audio. The tradeoff for better transient response is a higher bitrate. ### mjpeg Motion JPEG encoder. _Options_ ### huffman Set the huffman encoding strategy. Possible values: **default** Use the default huffman tables. This is the default strategy. **optimal** Compute and use optimal huffman tables. ### wavpack WavPack lossless audio encoder. _Options_ The equivalent options for **wavpack **command line utility are listed in parentheses. Shared options The following shared options are effective for this encoder. Only special notes about this particular encoder will be documented here. For the general meaning of the options, see **the** **Codec Options chapter**. **frame_size (_**--blocksize_**)** For this encoder, the range for this option is between 128 and 131072. Default is automatically decided based on sample rate and number of channel. For the complete formula of calculating default, see _libavcodec/wavpackenc.c_. **compression_level (_**-f_**, _**-h_**, _**-hh_**, and _**-x_**)** Private options **joint_stereo (_**-j_**)** Set whether to enable joint stereo. Valid values are: **on (_**1_**)** Force mid/side audio encoding. **off (_**0_**)** Force left/right audio encoding. **auto** Let the encoder decide automatically. ### optimize_mono Set whether to enable optimization for mono. This option is only effective for non-mono streams. Available values: **on **enabled **off **disabled ## VIDEO ENCODERS A description of some of the currently available video encoders follows. ### a64_multi, a64_multi5 A64 / Commodore 64 multicolor charset encoder. "a64_multi5" is extended with 5th color (colram). ### Cinepak Cinepak aka CVID encoder. Compatible with Windows 3.1 and vintage MacOS. _Options_ **g _**integer_ Keyframe interval. A keyframe is inserted at least every "-g" frames, sometimes sooner. **q:v _**integer_ Quality factor. Lower is better. Higher gives lower bitrate. The following table lists bitrates when encoding akiyo_cif.y4m for various values of "-q:v" with "-g 100": **"-q:v 1" 1918 kb/s** **"-q:v 2" 1735 kb/s** **"-q:v 4" 1500 kb/s** **"-q:v 10" 1041 kb/s** **"-q:v 20" 826 kb/s** **"-q:v 40" 553 kb/s** **"-q:v 100" 394 kb/s** **"-q:v 200" 312 kb/s** **"-q:v 400" 266 kb/s** **"-q:v 1000" 237 kb/s** **max_extra_cb_iterations _**integer_ Max extra codebook recalculation passes, more is better and slower. **skip_empty_cb _**boolean_ Avoid wasting bytes, ignore vintage MacOS decoder. **max_strips _**integer_ **min_strips _**integer_ The minimum and maximum number of strips to use. Wider range sometimes improves quality. More strips is generally better quality but costs more bits. Fewer strips tend to yield more keyframes. Vintage compatible is 1..3. **strip_number_adaptivity _**integer_ How much number of strips is allowed to change between frames. Higher is better but slower. **GIF** GIF image/animation encoder. _Options_ **gifflags _**integer_ Sets the flags used for GIF encoding. **offsetting** Enables picture offsetting. Default is enabled. **transdiff** Enables transparency detection between frames. Default is enabled. **gifimage _**integer_ Enables encoding one full GIF image per frame, rather than an animated GIF. Default value is **0**. **global_palette _**integer_ Writes a palette to the global GIF header where feasible. If disabled, every frame will always have a palette written, even if there is a global palette supplied. Default value is **1**. ### Hap Vidvox Hap video encoder. _Options_ **format _**integer_ Specifies the Hap format to encode. **hap** **hap_alpha** **hap_q** Default value is **hap**. **chunks _**integer_ Specifies the number of chunks to split frames into, between 1 and 64. This permits multithreaded decoding of large frames, potentially at the cost of data-rate. The encoder may modify this value to divide frames evenly. Default value is _1_. **compressor _**integer_ Specifies the second-stage compressor to use. If set to **none**, **chunks **will be limited to 1, as chunked uncompressed frames offer no benefit. **none** **snappy** Default value is **snappy**. ### jpeg2000 The native jpeg 2000 encoder is lossy by default, the "-q:v" option can be used to set the encoding quality. Lossless encoding can be selected with "-pred 1". _Options_ **format _**integer_ Can be set to either "j2k" or "jp2" (the default) that makes it possible to store non-rgb pix_fmts. **tile_width _**integer_ Sets tile width. Range is 1 to 1073741824. Default is 256. **tile_height _**integer_ Sets tile height. Range is 1 to 1073741824. Default is 256. **pred _**integer_ Allows setting the discrete wavelet transform (DWT) type **dwt97int (Lossy)** **dwt53 (Lossless)** Default is "dwt97int" **sop _**boolean_ Enable this to add SOP marker at the start of each packet. Disabled by default. **eph _**boolean_ Enable this to add EPH marker at the end of each packet header. Disabled by default. **prog _**integer_ Sets the progression order to be used by the encoder. Possible values are: **lrcp** **rlcp** **rpcl** **pcrl** **cprl** Set to "lrcp" by default. **layer_rates _**string_ By default, when this option is not used, compression is done using the quality metric. This option allows for compression using compression ratio. The compression ratio for each level could be specified. The compression ratio of a layer "l" species the what ratio of total file size is contained in the first "l" layers. Example usage: ffmpeg -i input.bmp -c:v jpeg2000 -layer_rates "100,10,1" output.j2k This would compress the image to contain 3 layers, where the data contained in the first layer would be compressed by 1000 times, compressed by 100 in the first two layers, and shall contain all data while using all 3 layers. ### librav1e rav1e AV1 encoder wrapper. Requires the presence of the rav1e headers and library during configuration. You need to explicitly configure the build with "--enable-librav1e". _Options_ ### qmax Sets the maximum quantizer to use when using bitrate mode. ### qmin Sets the minimum quantizer to use when using bitrate mode. **qp **Uses quantizer mode to encode at the given quantizer (0-255). ### speed Selects the speed preset (0-10) to encode with. ### tiles Selects how many tiles to encode with. ### tile-rows Selects how many rows of tiles to encode with. ### tile-columns Selects how many columns of tiles to encode with. ### rav1e-params Set rav1e options using a list of _key_=_value_ pairs separated by ":". See **rav1e --help **for a list of options. For example to specify librav1e encoding options with **-rav1e-params**: ffmpeg -i input -c:v librav1e -b:v 500K -rav1e-params speed=5:low_latency=true output.mp4 ### libaom-av1 libaom AV1 encoder wrapper. Requires the presence of the libaom headers and library during configuration. You need to explicitly configure the build with "--enable-libaom". _Options_ The wrapper supports the following standard libavcodec options: **b **Set bitrate target in bits/second. By default this will use variable-bitrate mode. If **maxrate **and **minrate **are also set to the same value then it will use constant-bitrate mode, otherwise if **crf **is set as well then it will use constrained-quality mode. ### g keyint_min Set key frame placement. The GOP size sets the maximum distance between key frames; if zero the output stream will be intra-only. The minimum distance is ignored unless it is the same as the GOP size, in which case key frames will always appear at a fixed interval. Not set by default, so without this option the library has completely free choice about where to place key frames. ### qmin qmax Set minimum/maximum quantisation values. Valid range is from 0 to 63 (warning: this does not match the quantiser values actually used by AV1 - divide by four to map real quantiser values to this range). Defaults to min/max (no constraint). ### minrate maxrate bufsize rc_init_occupancy Set rate control buffering parameters. Not used if not set - defaults to unconstrained variable bitrate. ### threads Set the number of threads to use while encoding. This may require the **tiles **or **row-mt** options to also be set to actually use the specified number of threads fully. Defaults to the number of hardware threads supported by the host machine. ### profile Set the encoding profile. Defaults to using the profile which matches the bit depth and chroma subsampling of the input. The wrapper also has some specific options: ### cpu-used Set the quality/encoding speed tradeoff. Valid range is from 0 to 8, higher numbers indicating greater speed and lower quality. The default value is 1, which will be slow and high quality. ### auto-alt-ref Enable use of alternate reference frames. Defaults to the internal default of the library. **arnr-max-frames (_**frames_**)** Set altref noise reduction max frame count. Default is -1. **arnr-strength (_**strength_**)** Set altref noise reduction filter strength. Range is -1 to 6. Default is -1. **aq-mode (_**aq-mode_**)** Set adaptive quantization mode. Possible values: **none (_**0_**)** Disabled. **variance (_**1_**)** Variance-based. **complexity (_**2_**)** Complexity-based. **cyclic (_**3_**)** Cyclic refresh. **tune (_**tune_**)** Set the distortion metric the encoder is tuned with. Default is "psnr". **psnr (_**0_**)** **ssim (_**1_**)** ### lag-in-frames Set the maximum number of frames which the encoder may keep in flight at any one time for lookahead purposes. Defaults to the internal default of the library. ### error-resilience Enable error resilience features: **default** Improve resilience against losses of whole frames. Not enabled by default. **crf **Set the quality/size tradeoff for constant-quality (no bitrate target) and constrained- quality (with maximum bitrate target) modes. Valid range is 0 to 63, higher numbers indicating lower quality and smaller output size. Only used if set; by default only the bitrate target is used. ### static-thresh Set a change threshold on blocks below which they will be skipped by the encoder. Defined in arbitrary units as a nonnegative integer, defaulting to zero (no blocks are skipped). ### drop-threshold Set a threshold for dropping frames when close to rate control bounds. Defined as a percentage of the target buffer - when the rate control buffer falls below this percentage, frames will be dropped until it has refilled above the threshold. Defaults to zero (no frames are dropped). **denoise-noise-level (_**level_**)** Amount of noise to be removed for grain synthesis. Grain synthesis is disabled if this option is not set or set to 0. **denoise-block-size (_**pixels_**)** Block size used for denoising for grain synthesis. If not set, AV1 codec uses the default value of 32. **undershoot-pct (_**pct_**)** Set datarate undershoot (min) percentage of the target bitrate. Range is -1 to 100. Default is -1. **overshoot-pct (_**pct_**)** Set datarate overshoot (max) percentage of the target bitrate. Range is -1 to 1000. Default is -1. **minsection-pct (_**pct_**)** Minimum percentage variation of the GOP bitrate from the target bitrate. If minsection- pct is not set, the libaomenc wrapper computes it as follows: "(minrate * 100 / bitrate)". Range is -1 to 100. Default is -1 (unset). **maxsection-pct (_**pct_**)** Maximum percentage variation of the GOP bitrate from the target bitrate. If maxsection- pct is not set, the libaomenc wrapper computes it as follows: "(maxrate * 100 / bitrate)". Range is -1 to 5000. Default is -1 (unset). **frame-parallel (_**boolean_**)** Enable frame parallel decodability features. Default is true. ### tiles Set the number of tiles to encode the input video with, as columns x rows. Larger numbers allow greater parallelism in both encoding and decoding, but may decrease coding efficiency. Defaults to the minimum number of tiles required by the size of the input video (this is 1x1 (that is, a single tile) for sizes up to and including 4K). ### tile-columns tile-rows Set the number of tiles as log2 of the number of tile rows and columns. Provided for compatibility with libvpx/VP9. ### row-mt (Requires libaom >= 1.0.0-759-g90a15f4f2) Enable row based multi-threading. Disabled by default. **enable-cdef (_**boolean_**)** Enable Constrained Directional Enhancement Filter. The libaom-av1 encoder enables CDEF by default. **enable-restoration (_**boolean_**)** Enable Loop Restoration Filter. Default is true for libaom-av1. **enable-global-motion (_**boolean_**)** Enable the use of global motion for block prediction. Default is true. **enable-intrabc (_**boolean_**)** Enable block copy mode for intra block prediction. This mode is useful for screen content. Default is true. **enable-rect-partitions (_**boolean_**) (Requires libaom >= v2.0.0)** Enable rectangular partitions. Default is true. **enable-1to4-partitions (_**boolean_**) (Requires libaom >= v2.0.0)** Enable 1:4/4:1 partitions. Default is true. **enable-ab-partitions (_**boolean_**) (Requires libaom >= v2.0.0)** Enable AB shape partitions. Default is true. **enable-angle-delta (_**boolean_**) (Requires libaom >= v2.0.0)** Enable angle delta intra prediction. Default is true. **enable-cfl-intra (_**boolean_**) (Requires libaom >= v2.0.0)** Enable chroma predicted from luma intra prediction. Default is true. **enable-filter-intra (_**boolean_**) (Requires libaom >= v2.0.0)** Enable filter intra predictor. Default is true. **enable-intra-edge-filter (_**boolean_**) (Requires libaom >= v2.0.0)** Enable intra edge filter. Default is true. **enable-smooth-intra (_**boolean_**) (Requires libaom >= v2.0.0)** Enable smooth intra prediction mode. Default is true. **enable-paeth-intra (_**boolean_**) (Requires libaom >= v2.0.0)** Enable paeth predictor in intra prediction. Default is true. **enable-palette (_**boolean_**) (Requires libaom >= v2.0.0)** Enable palette prediction mode. Default is true. **enable-flip-idtx (_**boolean_**) (Requires libaom >= v2.0.0)** Enable extended transform type, including FLIPADST_DCT, DCT_FLIPADST, FLIPADST_FLIPADST, ADST_FLIPADST, FLIPADST_ADST, IDTX, V_DCT, H_DCT, V_ADST, H_ADST, V_FLIPADST, H_FLIPADST. Default is true. **enable-tx64 (_**boolean_**) (Requires libaom >= v2.0.0)** Enable 64-pt transform. Default is true. **reduced-tx-type-set (_**boolean_**) (Requires libaom >= v2.0.0)** Use reduced set of transform types. Default is false. **use-intra-dct-only (_**boolean_**) (Requires libaom >= v2.0.0)** Use DCT only for INTRA modes. Default is false. **use-inter-dct-only (_**boolean_**) (Requires libaom >= v2.0.0)** Use DCT only for INTER modes. Default is false. **use-intra-default-tx-only (_**boolean_**) (Requires libaom >= v2.0.0)** Use Default-transform only for INTRA modes. Default is false. **enable-ref-frame-mvs (_**boolean_**) (Requires libaom >= v2.0.0)** Enable temporal mv prediction. Default is true. **enable-reduced-reference-set (_**boolean_**) (Requires libaom >= v2.0.0)** Use reduced set of single and compound references. Default is false. **enable-obmc (_**boolean_**) (Requires libaom >= v2.0.0)** Enable obmc. Default is true. **enable-dual-filter (_**boolean_**) (Requires libaom >= v2.0.0)** Enable dual filter. Default is true. **enable-diff-wtd-comp (_**boolean_**) (Requires libaom >= v2.0.0)** Enable difference-weighted compound. Default is true. **enable-dist-wtd-comp (_**boolean_**) (Requires libaom >= v2.0.0)** Enable distance-weighted compound. Default is true. **enable-onesided-comp (_**boolean_**) (Requires libaom >= v2.0.0)** Enable one sided compound. Default is true. **enable-interinter-wedge (_**boolean_**) (Requires libaom >= v2.0.0)** Enable interinter wedge compound. Default is true. **enable-interintra-wedge (_**boolean_**) (Requires libaom >= v2.0.0)** Enable interintra wedge compound. Default is true. **enable-masked-comp (_**boolean_**) (Requires libaom >= v2.0.0)** Enable masked compound. Default is true. **enable-interintra-comp (_**boolean_**) (Requires libaom >= v2.0.0)** Enable interintra compound. Default is true. **enable-smooth-interintra (_**boolean_**) (Requires libaom >= v2.0.0)** Enable smooth interintra mode. Default is true. ### aom-params Set libaom options using a list of _key_=_value_ pairs separated by ":". For a list of supported options, see **aomenc --help **under the section "AV1 Specific Options". For example to specify libaom encoding options with **-aom-params**: ffmpeg -i input -c:v libaom-av1 -b:v 500K -aom-params tune=psnr:enable-tpl-model=1 output.mp4 ### libsvtav1 SVT-AV1 encoder wrapper. Requires the presence of the SVT-AV1 headers and library during configuration. You need to explicitly configure the build with "--enable-libsvtav1". _Options_ ### profile Set the encoding profile. **main** **high** **professional** ### level Set the operating point level. For example: '4.0' ### hielevel Set the Hierarchical prediction levels. **3level** **4level** This is the default. ### tier Set the operating point tier. **main** This is the default. **high** ### qmax Set the maximum quantizer to use when using a bitrate mode. ### qmin Set the minimum quantizer to use when using a bitrate mode. **crf **Constant rate factor value used in crf rate control mode (0-63). **qp **Set the quantizer used in cqp rate control mode (0-63). ### sc_detection Enable scene change detection. ### la_depth Set number of frames to look ahead (0-120). ### preset Set the quality-speed tradeoff, in the range 0 to 13. Higher values are faster but lower quality. ### tile_rows Set log2 of the number of rows of tiles to use (0-6). ### tile_columns Set log2 of the number of columns of tiles to use (0-4). ### svtav1-params Set SVT-AV1 options using a list of _key_=_value_ pairs separated by ":". See the SVT-AV1 encoder user guide for a list of accepted parameters. ### libjxl libjxl JPEG XL encoder wrapper. Requires the presence of the libjxl headers and library during configuration. You need to explicitly configure the build with "--enable-libjxl". _Options_ The libjxl wrapper supports the following options: ### distance Set the target Butteraugli distance. This is a quality setting: lower distance yields higher quality, with distance=1.0 roughly comparable to libjpeg Quality 90 for photographic content. Setting distance=0.0 yields true lossless encoding. Valid values range between 0.0 and 15.0, and sane values rarely exceed 5.0. Setting distance=0.1 usually attains transparency for most input. The default is 1.0. ### effort Set the encoding effort used. Higher effort values produce more consistent quality and usually produces a better quality/bpp curve, at the cost of more CPU time required. Valid values range from 1 to 9, and the default is 7. ### modular Force the encoder to use Modular mode instead of choosing automatically. The default is to use VarDCT for lossy encoding and Modular for lossless. VarDCT is generally superior to Modular for lossy encoding but does not support lossless encoding. ### libkvazaar Kvazaar H.265/HEVC encoder. Requires the presence of the libkvazaar headers and library during configuration. You need to explicitly configure the build with **--enable-libkvazaar**. _Options_ **b **Set target video bitrate in bit/s and enable rate control. ### kvazaar-params Set kvazaar parameters as a list of _name_=_value_ pairs separated by commas (,). See kvazaar documentation for a list of options. ### libopenh264 Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper. This encoder requires the presence of the libopenh264 headers and library during configuration. You need to explicitly configure the build with "--enable-libopenh264". The library is detected using **pkg-config**. For more information about the library see <****>. _Options_ The following FFmpeg global options affect the configurations of the libopenh264 encoder. **b **Set the bitrate (as a number of bits per second). **g **Set the GOP size. ### maxrate Set the max bitrate (as a number of bits per second). ### flags +global_header Set global header in the bitstream. ### slices Set the number of slices, used in parallelized encoding. Default value is 0. This is only used when **slice_mode **is set to **fixed**. ### loopfilter Enable loop filter, if set to 1 (automatically enabled). To disable set a value of 0. ### profile Set profile restrictions. If set to the value of **main **enable CABAC (set the "SEncParamExt.iEntropyCodingModeFlag" flag to 1). ### max_nal_size Set maximum NAL size in bytes. ### allow_skip_frames Allow skipping frames to hit the target bitrate if set to 1. ### libtheora libtheora Theora encoder wrapper. Requires the presence of the libtheora headers and library during configuration. You need to explicitly configure the build with "--enable-libtheora". For more information about the libtheora project see <****>. _Options_ The following global options are mapped to internal libtheora options which affect the quality and the bitrate of the encoded stream. **b **Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In case VBR (Variable Bit Rate) mode is enabled this option is ignored. ### flags Used to enable constant quality mode (VBR) encoding through the **qscale **flag, and to enable the "pass1" and "pass2" modes. **g **Set the GOP size. ### global_quality Set the global quality as an integer in lambda units. Only relevant when VBR mode is enabled with "flags +qscale". The value is converted to QP units by dividing it by "FF_QP2LAMBDA", clipped in the [0 - 10] range, and then multiplied by 6.3 to get a value in the native libtheora range [0-63]. A higher value corresponds to a higher quality. **q **Enable VBR mode when set to a non-negative value, and set constant quality value as a double floating point value in QP units. The value is clipped in the [0-10] range, and then multiplied by 6.3 to get a value in the native libtheora range [0-63]. This option is valid only using the **ffmpeg **command-line tool. For library interface users, use **global_quality**. _Examples_ • Set maximum constant quality (VBR) encoding with **ffmpeg**: ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg • Use **ffmpeg **to convert a CBR 1000 kbps Theora video stream: ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg ### libvpx VP8/VP9 format supported through libvpx. Requires the presence of the libvpx headers and library during configuration. You need to explicitly configure the build with "--enable-libvpx". _Options_ The following options are supported by the libvpx wrapper. The **vpxenc**-equivalent options or values are listed in parentheses for easy migration. To reduce the duplication of documentation, only the private options and some others requiring special attention are documented here. For the documentation of the undocumented generic options, see **the Codec Options chapter**. To get more documentation of the libvpx options, invoke the command **ffmpeg -h encoder=libvpx**, **ffmpeg -h encoder=libvpx-vp9 **or **vpxenc --help**. Further information is available in the libvpx API documentation. **b (_**target-bitrate_**)** Set bitrate in bits/s. Note that FFmpeg's **b **option is expressed in bits/s, while **vpxenc**'s **target-bitrate **is in kilobits/s. **g (_**kf-max-dist_**)** **keyint_min (_**kf-min-dist_**)** **qmin (_**min-q_**)** Minimum (Best Quality) Quantizer. **qmax (_**max-q_**)** Maximum (Worst Quality) Quantizer. Can be changed per-frame. **bufsize (_**buf-sz_**, _**buf-optimal-sz_**)** Set ratecontrol buffer size (in bits). Note **vpxenc**'s options are specified in milliseconds, the libvpx wrapper converts this value as follows: "buf-sz = bufsize * 1000 / bitrate", "buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6". **rc_init_occupancy (_**buf-initial-sz_**)** Set number of bits which should be loaded into the rc buffer before decoding starts. Note **vpxenc**'s option is specified in milliseconds, the libvpx wrapper converts this value as follows: "rc_init_occupancy * 1000 / bitrate". ### undershoot-pct Set datarate undershoot (min) percentage of the target bitrate. ### overshoot-pct Set datarate overshoot (max) percentage of the target bitrate. **skip_threshold (_**drop-frame_**)** **qcomp (_**bias-pct_**)** **maxrate (_**maxsection-pct_**)** Set GOP max bitrate in bits/s. Note **vpxenc**'s option is specified as a percentage of the target bitrate, the libvpx wrapper converts this value as follows: "(maxrate * 100 / bitrate)". **minrate (_**minsection-pct_**)** Set GOP min bitrate in bits/s. Note **vpxenc**'s option is specified as a percentage of the target bitrate, the libvpx wrapper converts this value as follows: "(minrate * 100 / bitrate)". **minrate, maxrate, b _**end-usage=cbr_ "(minrate == maxrate == bitrate)". **crf (_**end-usage=cq_**, _**cq-level_**)** **tune (_**tune_**)** **psnr (_**psnr_**)** **ssim (_**ssim_**)** **quality, deadline (_**deadline_**)** **best** Use best quality deadline. Poorly named and quite slow, this option should be avoided as it may give worse quality output than good. **good** Use good quality deadline. This is a good trade-off between speed and quality when used with the **cpu-used **option. **realtime** Use realtime quality deadline. **speed, cpu-used (_**cpu-used_**)** Set quality/speed ratio modifier. Higher values speed up the encode at the cost of quality. **nr (_**noise-sensitivity_**)** ### static-thresh Set a change threshold on blocks below which they will be skipped by the encoder. **slices (_**token-parts_**)** Note that FFmpeg's **slices **option gives the total number of partitions, while **vpxenc**'s **token-parts **is given as log2(partitions). ### max-intra-rate Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0 means unlimited. ### force_key_frames "VPX_EFLAG_FORCE_KF" ### Alternate reference frame related **auto-alt-ref** Enable use of alternate reference frames (2-pass only). Values greater than 1 enable multi-layer alternate reference frames (VP9 only). **arnr-maxframes** Set altref noise reduction max frame count. **arnr-type** Set altref noise reduction filter type: backward, forward, centered. **arnr-strength** Set altref noise reduction filter strength. **rc-lookahead, lag-in-frames (_**lag-in-frames_**)** Set number of frames to look ahead for frametype and ratecontrol. **min-gf-interval** Set minimum golden/alternate reference frame interval (VP9 only). ### error-resilient Enable error resiliency features. **sharpness _**integer_ Increase sharpness at the expense of lower PSNR. The valid range is [0, 7]. ### ts-parameters Sets the temporal scalability configuration using a :-separated list of key=value pairs. For example, to specify temporal scalability parameters with "ffmpeg": ffmpeg -i INPUT -c:v libvpx -ts-parameters ts_number_layers=3:\ ts_target_bitrate=250,500,1000:ts_rate_decimator=4,2,1:\ ts_periodicity=4:ts_layer_id=0,2,1,2:ts_layering_mode=3 OUTPUT Below is a brief explanation of each of the parameters, please refer to "struct vpx_codec_enc_cfg" in "vpx/vpx_encoder.h" for more details. **ts_number_layers** Number of temporal coding layers. **ts_target_bitrate** Target bitrate for each temporal layer (in kbps). (bitrate should be inclusive of the lower temporal layer). **ts_rate_decimator** Frame rate decimation factor for each temporal layer. **ts_periodicity** Length of the sequence defining frame temporal layer membership. **ts_layer_id** Template defining the membership of frames to temporal layers. **ts_layering_mode** (optional) Selecting the temporal structure from a set of pre-defined temporal layering modes. Currently supports the following options. **0 **No temporal layering flags are provided internally, relies on flags being passed in using "metadata" field in "AVFrame" with following keys. **vp8-flags** Sets the flags passed into the encoder to indicate the referencing scheme for the current frame. Refer to function "vpx_codec_encode" in "vpx/vpx_encoder.h" for more details. **temporal_id** Explicitly sets the temporal id of the current frame to encode. **2 **Two temporal layers. 0-1... **3 **Three temporal layers. 0-2-1-2...; with single reference frame. **4 **Same as option "3", except there is a dependency between the two temporal layer 2 frames within the temporal period. ### VP9-specific options **lossless** Enable lossless mode. **tile-columns** Set number of tile columns to use. Note this is given as log2(tile_columns). For example, 8 tile columns would be requested by setting the **tile-columns **option to 3. **tile-rows** Set number of tile rows to use. Note this is given as log2(tile_rows). For example, 4 tile rows would be requested by setting the **tile-rows **option to 2. **frame-parallel** Enable frame parallel decodability features. **aq-mode** Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3: cyclic refresh, 4: equator360). **colorspace _**color-space_ Set input color space. The VP9 bitstream supports signaling the following colorspaces: **rgb sRGB** **bt709 bt709** **unspecified unknown** **bt470bg bt601** **smpte170m smpte170** **smpte240m smpte240** **bt2020_ncl bt2020** **row-mt _**boolean_ Enable row based multi-threading. **tune-content** Set content type: default (0), screen (1), film (2). **corpus-complexity** Corpus VBR mode is a variant of standard VBR where the complexity distribution midpoint is passed in rather than calculated for a specific clip or chunk. The valid range is [0, 10000]. 0 (default) uses standard VBR. **enable-tpl _**boolean_ Enable temporal dependency model. **ref-frame-config** Using per-frame metadata, set members of the structure "vpx_svc_ref_frame_config_t" in "vpx/vp8cx.h" to fine-control referencing schemes and frame buffer management. Use a :-separated list of key=value pairs. For example, av_dict_set(&av_frame->metadata, "ref-frame-config", \ "rfc_update_buffer_slot=7:rfc_lst_fb_idx=0:rfc_gld_fb_idx=1:rfc_alt_fb_idx=2:rfc_reference_last=0:rfc_reference_golden=0:rfc_reference_alt_ref=0"); **rfc_update_buffer_slot** Indicates the buffer slot number to update **rfc_update_last** Indicates whether to update the LAST frame **rfc_update_golden** Indicates whether to update GOLDEN frame **rfc_update_alt_ref** Indicates whether to update ALT_REF frame **rfc_lst_fb_idx** LAST frame buffer index **rfc_gld_fb_idx** GOLDEN frame buffer index **rfc_alt_fb_idx** ALT_REF frame buffer index **rfc_reference_last** Indicates whether to reference LAST frame **rfc_reference_golden** Indicates whether to reference GOLDEN frame **rfc_reference_alt_ref** Indicates whether to reference ALT_REF frame **rfc_reference_duration** Indicates frame duration For more information about libvpx see: <****> ### libwebp libwebp WebP Image encoder wrapper libwebp is Google's official encoder for WebP images. It can encode in either lossy or lossless mode. Lossy images are essentially a wrapper around a VP8 frame. Lossless images are a separate codec developed by Google. _Pixel_ _Format_ Currently, libwebp only supports YUV420 for lossy and RGB for lossless due to limitations of the format and libwebp. Alpha is supported for either mode. Because of API limitations, if RGB is passed in when encoding lossy or YUV is passed in for encoding lossless, the pixel format will automatically be converted using functions from libwebp. This is not ideal and is done only for convenience. _Options_ ### -lossless _ Enables/Disables use of lossless mode. Default is 0. ### -compression_level _ For lossy, this is a quality/speed tradeoff. Higher values give better quality for a given size at the cost of increased encoding time. For lossless, this is a size/speed tradeoff. Higher values give smaller size at the cost of increased encoding time. More specifically, it controls the number of extra algorithms and compression tools used, and varies the combination of these tools. This maps to the _method_ option in libwebp. The valid range is 0 to 6. Default is 4. ### -quality _ For lossy encoding, this controls image quality. For lossless encoding, this controls the effort and time spent in compression. Range is 0 to 100. Default is 75. ### -preset _ Configuration preset. This does some automatic settings based on the general type of the image. **none** Do not use a preset. **default** Use the encoder default. **picture** Digital picture, like portrait, inner shot **photo** Outdoor photograph, with natural lighting **drawing** Hand or line drawing, with high-contrast details **icon** Small-sized colorful images **text** Text-like ### libx264, libx264rgb x264 H.264/MPEG-4 AVC encoder wrapper. This encoder requires the presence of the libx264 headers and library during configuration. You need to explicitly configure the build with "--enable-libx264". libx264 supports an impressive number of features, including 8x8 and 4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC entropy coding, interlacing (MBAFF), lossless mode, psy optimizations for detail retention (adaptive quantization, psy-RD, psy- trellis). Many libx264 encoder options are mapped to FFmpeg global codec options, while unique encoder options are provided through private options. Additionally the **x264opts **and **x264-params** private options allows one to pass a list of key=value tuples as accepted by the libx264 "x264_param_parse" function. The x264 project website is at <****>. The libx264rgb encoder is the same as libx264, except it accepts packed RGB pixel formats as input instead of YUV. _Supported_ _Pixel_ _Formats_ x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at x264's configure time. _Options_ The following options are supported by the libx264 wrapper. The **x264**-equivalent options or values are listed in parentheses for easy migration. To reduce the duplication of documentation, only the private options and some others requiring special attention are documented here. For the documentation of the undocumented generic options, see **the Codec Options chapter**. To get a more accurate and extensive documentation of the libx264 options, invoke the command **x264 --fullhelp **or consult the libx264 documentation. **b (_**bitrate_**)** Set bitrate in bits/s. Note that FFmpeg's **b **option is expressed in bits/s, while **x264**'s **bitrate **is in kilobits/s. **bf (_**bframes_**)** **g (_**keyint_**)** **qmin (_**qpmin_**)** Minimum quantizer scale. **qmax (_**qpmax_**)** Maximum quantizer scale. **qdiff (_**qpstep_**)** Maximum difference between quantizer scales. **qblur (_**qblur_**)** Quantizer curve blur **qcomp (_**qcomp_**)** Quantizer curve compression factor **refs (_**ref_**)** Number of reference frames each P-frame can use. The range is from _0-16_. **level (_**level_**)** Set the "x264_param_t.i_level_idc" value in case the value is positive, it is ignored otherwise. This value can be set using the "AVCodecContext" API (e.g. by setting the "AVCodecContext" value directly), and is specified as an integer mapped on a corresponding level (e.g. the value 31 maps to H.264 level IDC "3.1", as defined in the "x264_levels" table). It is ignored when set to a non positive value. Alternatively it can be set as a private option, overriding the value set in "AVCodecContext", and in this case must be specified as the level IDC identifier (e.g. "3.1"), as defined by H.264 Annex A. **sc_threshold (_**scenecut_**)** Sets the threshold for the scene change detection. **trellis (_**trellis_**)** Performs Trellis quantization to increase efficiency. Enabled by default. **nr (_**nr_**)** **me_range (_**merange_**)** Maximum range of the motion search in pixels. **me_method (_**me_**)** Set motion estimation method. Possible values in the decreasing order of speed: **dia (_**dia_**)** **epzs (_**dia_**)** Diamond search with radius 1 (fastest). **epzs **is an alias for **dia**. **hex (_**hex_**)** Hexagonal search with radius 2. **umh (_**umh_**)** Uneven multi-hexagon search. **esa (_**esa_**)** Exhaustive search. **tesa (_**tesa_**)** Hadamard exhaustive search (slowest). ### forced-idr Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces it to choose an IDR-frame. **subq (_**subme_**)** Sub-pixel motion estimation method. **b_strategy (_**b-adapt_**)** Adaptive B-frame placement decision algorithm. Use only on first-pass. **keyint_min (_**min-keyint_**)** Minimum GOP size. ### coder Set entropy encoder. Possible values: **ac **Enable CABAC. **vlc **Enable CAVLC and disable CABAC. It generates the same effect as **x264**'s **--no-cabac** option. **cmp **Set full pixel motion estimation comparison algorithm. Possible values: **chroma** Enable chroma in motion estimation. **sad **Ignore chroma in motion estimation. It generates the same effect as **x264**'s **--no-chroma-me **option. **threads (_**threads_**)** Number of encoding threads. ### thread_type Set multithreading technique. Possible values: **slice** Slice-based multithreading. It generates the same effect as **x264**'s **--sliced-threads** option. **frame** Frame-based multithreading. ### flags Set encoding flags. It can be used to disable closed GOP and enable open GOP by setting it to "-cgop". The result is similar to the behavior of **x264**'s **--open-gop **option. **rc_init_occupancy (_**vbv-init_**)** **preset (_**preset_**)** Set the encoding preset. **tune (_**tune_**)** Set tuning of the encoding params. **profile (_**profile_**)** Set profile restrictions. ### fastfirstpass Enable fast settings when encoding first pass, when set to 1. When set to 0, it has the same effect of **x264**'s **--slow-firstpass **option. **crf (_**crf_**)** Set the quality for constant quality mode. **crf_max (_**crf-max_**)** In CRF mode, prevents VBV from lowering quality beyond this point. **qp (_**qp_**)** Set constant quantization rate control method parameter. **aq-mode (_**aq-mode_**)** Set AQ method. Possible values: **none (_**0_**)** Disabled. **variance (_**1_**)** Variance AQ (complexity mask). **autovariance (_**2_**)** Auto-variance AQ (experimental). **aq-strength (_**aq-strength_**)** Set AQ strength, reduce blocking and blurring in flat and textured areas. **psy **Use psychovisual optimizations when set to 1. When set to 0, it has the same effect as **x264**'s **--no-psy **option. **psy-rd (_**psy-rd_**)** Set strength of psychovisual optimization, in _psy-rd_:_psy-trellis_ format. **rc-lookahead (_**rc-lookahead_**)** Set number of frames to look ahead for frametype and ratecontrol. ### weightb Enable weighted prediction for B-frames when set to 1. When set to 0, it has the same effect as **x264**'s **--no-weightb **option. **weightp (_**weightp_**)** Set weighted prediction method for P-frames. Possible values: **none (_**0_**)** Disabled **simple (_**1_**)** Enable only weighted refs **smart (_**2_**)** Enable both weighted refs and duplicates **ssim (_**ssim_**)** Enable calculation and printing SSIM stats after the encoding. **intra-refresh (_**intra-refresh_**)** Enable the use of Periodic Intra Refresh instead of IDR frames when set to 1. **avcintra-class (_**class_**)** Configure the encoder to generate AVC-Intra. Valid values are 50,100 and 200 **bluray-compat (_**bluray-compat_**)** Configure the encoder to be compatible with the bluray standard. It is a shorthand for setting "bluray-compat=1 force-cfr=1". **b-bias (_**b-bias_**)** Set the influence on how often B-frames are used. **b-pyramid (_**b-pyramid_**)** Set method for keeping of some B-frames as references. Possible values: **none (_**none_**)** Disabled. **strict (_**strict_**)** Strictly hierarchical pyramid. **normal (_**normal_**)** Non-strict (not Blu-ray compatible). ### mixed-refs Enable the use of one reference per partition, as opposed to one reference per macroblock when set to 1. When set to 0, it has the same effect as **x264**'s **--no-mixed-refs **option. ### 8x8dct Enable adaptive spatial transform (high profile 8x8 transform) when set to 1. When set to 0, it has the same effect as **x264**'s **--no-8x8dct **option. ### fast-pskip Enable early SKIP detection on P-frames when set to 1. When set to 0, it has the same effect as **x264**'s **--no-fast-pskip **option. **aud (_**aud_**)** Enable use of access unit delimiters when set to 1. ### mbtree Enable use macroblock tree ratecontrol when set to 1. When set to 0, it has the same effect as **x264**'s **--no-mbtree **option. **deblock (_**deblock_**)** Set loop filter parameters, in _alpha_:_beta_ form. **cplxblur (_**cplxblur_**)** Set fluctuations reduction in QP (before curve compression). **partitions (_**partitions_**)** Set partitions to consider as a comma-separated list of. Possible values in the list: **p8x8** 8x8 P-frame partition. **p4x4** 4x4 P-frame partition. **b8x8** 4x4 B-frame partition. **i8x8** 8x8 I-frame partition. **i4x4** 4x4 I-frame partition. (Enabling **p4x4 **requires **p8x8 **to be enabled. Enabling **i8x8** requires adaptive spatial transform (**8x8dct **option) to be enabled.) **none (_**none_**)** Do not consider any partitions. **all (_**all_**)** Consider every partition. **direct-pred (_**direct_**)** Set direct MV prediction mode. Possible values: **none (_**none_**)** Disable MV prediction. **spatial (_**spatial_**)** Enable spatial predicting. **temporal (_**temporal_**)** Enable temporal predicting. **auto (_**auto_**)** Automatically decided. **slice-max-size (_**slice-max-size_**)** Set the limit of the size of each slice in bytes. If not specified but RTP payload size (**ps**) is specified, that is used. **stats (_**stats_**)** Set the file name for multi-pass stats. **nal-hrd (_**nal-hrd_**)** Set signal HRD information (requires **vbv-bufsize **to be set). Possible values: **none (_**none_**)** Disable HRD information signaling. **vbr (_**vbr_**)** Variable bit rate. **cbr (_**cbr_**)** Constant bit rate (not allowed in MP4 container). ### x264opts (N.A.) Set any x264 option, see **x264 --fullhelp **for a list. Argument is a list of _key_=_value_ couples separated by ":". In _filter_ and _psy-rd_ options that use ":" as a separator themselves, use "," instead. They accept it as well since long ago but this is kept undocumented for some reason. For example to specify libx264 encoding options with **ffmpeg**: ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv **a53cc _**boolean_ Import closed captions (which must be ATSC compatible format) into output. Only the mpeg2 and h264 decoders provide these. Default is 1 (on). **udu_sei _**boolean_ Import user data unregistered SEI if available into output. Default is 0 (off). **mb_info _**boolean_ Set mb_info data through AVFrameSideData, only useful when used from the API. Default is 0 (off). ### x264-params (N.A.) Override the x264 configuration using a :-separated list of key=value parameters. This option is functionally the same as the **x264opts**, but is duplicated for compatibility with the Libav fork. For example to specify libx264 encoding options with **ffmpeg**: ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\ cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\ no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT Encoding ffpresets for common usages are provided so they can be used with the general presets system (e.g. passing the **pre **option). ### libx265 x265 H.265/HEVC encoder wrapper. This encoder requires the presence of the libx265 headers and library during configuration. You need to explicitly configure the build with **--enable-libx265**. _Options_ **b **Sets target video bitrate. **bf** **g **Set the GOP size. ### keyint_min Minimum GOP size. ### refs Number of reference frames each P-frame can use. The range is from _1-16_. ### preset Set the x265 preset. ### tune Set the x265 tune parameter. ### profile Set profile restrictions. **crf **Set the quality for constant quality mode. **qp **Set constant quantization rate control method parameter. ### qmin Minimum quantizer scale. ### qmax Maximum quantizer scale. ### qdiff Maximum difference between quantizer scales. ### qblur Quantizer curve blur ### qcomp Quantizer curve compression factor ### i_qfactor ### b_qfactor ### forced-idr Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces it to choose an IDR-frame. **udu_sei _**boolean_ Import user data unregistered SEI if available into output. Default is 0 (off). ### x265-params Set x265 options using a list of _key_=_value_ couples separated by ":". See **x265 --help **for a list of options. For example to specify libx265 encoding options with **-x265-params**: ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4 ### libxavs2 xavs2 AVS2-P2/IEEE1857.4 encoder wrapper. This encoder requires the presence of the libxavs2 headers and library during configuration. You need to explicitly configure the build with **--enable-libxavs2**. The following standard libavcodec options are used: • **b **/ **bit_rate** • **g **/ **gop_size** • **bf **/ **max_b_frames** The encoder also has its own specific options: _Options_ ### lcu_row_threads Set the number of parallel threads for rows from 1 to 8 (default 5). ### initial_qp Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set the initial qp for the first frame. **qp **Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set the qp value under constant-QP mode. ### max_qp Set the max qp for rate control from 1 to 63 (default 55). ### min_qp Set the min qp for rate control from 1 to 63 (default 20). ### speed_level Set the Speed level from 0 to 9 (default 0). Higher is better but slower. ### log_level Set the log level from -1 to 3 (default 0). -1: none, 0: error, 1: warning, 2: info, 3: debug. ### xavs2-params Set xavs2 options using a list of _key_=_value_ couples separated by ":". For example to specify libxavs2 encoding options with **-xavs2-params**: ffmpeg -i input -c:v libxavs2 -xavs2-params RdoqLevel=0 output.avs2 ### libxvid Xvid MPEG-4 Part 2 encoder wrapper. This encoder requires the presence of the libxvidcore headers and library during configuration. You need to explicitly configure the build with "--enable-libxvid --enable-gpl". The native "mpeg4" encoder supports the MPEG-4 Part 2 format, so users can encode to this format without this library. _Options_ The following options are supported by the libxvid wrapper. Some of the following options are listed but are not documented, and correspond to shared codec options. See **the Codec Options** **chapter **for their documentation. The other shared options which are not listed have no effect for the libxvid encoder. **b** **g** ### qmin ### qmax ### mpeg_quant ### threads **bf** ### b_qfactor ### b_qoffset ### flags Set specific encoding flags. Possible values: **mv4 **Use four motion vector by macroblock. **aic **Enable high quality AC prediction. **gray** Only encode grayscale. **gmc **Enable the use of global motion compensation (GMC). **qpel** Enable quarter-pixel motion compensation. **cgop** Enable closed GOP. **global_header** Place global headers in extradata instead of every keyframe. ### trellis ### me_quality Set motion estimation quality level. Possible values in decreasing order of speed and increasing order of quality: **0 **Use no motion estimation (default). **1, 2** Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks. **3, 4** Enable all of the things described above, plus advanced diamond zonal search for 8x8 blocks and half-pixel refinement for 8x8 blocks, also enable motion estimation on chroma planes for P and B-frames. **5, 6** Enable all of the things described above, plus extended 16x16 and 8x8 blocks search. **mbd **Set macroblock decision algorithm. Possible values in the increasing order of quality: **simple** Use macroblock comparing function algorithm (default). **bits** Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks. **rd **Enable all of the things described above, plus rate distortion-based half pixel and quarter pixel refinement for 8x8 blocks, and rate distortion-based search using square pattern. ### lumi_aq Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled). ### variance_aq Enable variance adaptive quantization when set to 1. Default is 0 (disabled). When combined with **lumi_aq**, the resulting quality will not be better than any of the two specified individually. In other words, the resulting quality will be the worse one of the two effects. ### ssim Set structural similarity (SSIM) displaying method. Possible values: **off **Disable displaying of SSIM information. **avg **Output average SSIM at the end of encoding to stdout. The format of showing the average SSIM is: Average SSIM: %f For users who are not familiar with C, %f means a float number, or a decimal (e.g. 0.939232). **frame** Output both per-frame SSIM data during encoding and average SSIM at the end of encoding to stdout. The format of per-frame information is: SSIM: avg: %1.3f min: %1.3f max: %1.3f For users who are not familiar with C, %1.3f means a float number rounded to 3 digits after the dot (e.g. 0.932). ### ssim_acc Set SSIM accuracy. Valid options are integers within the range of 0-4, while 0 gives the most accurate result and 4 computes the fastest. ### MediaFoundation This provides wrappers to encoders (both audio and video) in the MediaFoundation framework. It can access both SW and HW encoders. Video encoders can take input in either of nv12 or yuv420p form (some encoders support both, some support only either - in practice, nv12 is the safer choice, especially among HW encoders). ### Microsoft RLE Microsoft RLE aka MSRLE encoder. Only 8-bit palette mode supported. Compatible with Windows 3.1 and Windows 95. _Options_ **g _**integer_ Keyframe interval. A keyframe is inserted at least every "-g" frames, sometimes sooner. ### mpeg2 MPEG-2 video encoder. _Options_ ### profile Select the mpeg2 profile to encode: **422** **high** **ss **Spatially Scalable **snr **SNR Scalable **main** **simple** ### level Select the mpeg2 level to encode: **high** **high1440** **main** **low** **seq_disp_ext _**integer_ Specifies if the encoder should write a sequence_display_extension to the output. **-1** **auto** Decide automatically to write it or not (this is the default) by checking if the data to be written is different from the default or unspecified values. **0** **never** Never write it. **1** **always** Always write it. **video_format _**integer_ Specifies the video_format written into the sequence display extension indicating the source of the video pictures. The default is **unspecified**, can be **component**, **pal**, **ntsc**, **secam **or **mac**. For maximum compatibility, use **component**. **a53cc _**boolean_ Import closed captions (which must be ATSC compatible format) into output. Default is 1 (on). ### png PNG image encoder. _Private_ _options_ **dpi _**integer_ Set physical density of pixels, in dots per inch, unset by default **dpm _**integer_ Set physical density of pixels, in dots per meter, unset by default ### ProRes Apple ProRes encoder. FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder. The used encoder can be chosen with the "-vcodec" option. _Private_ _Options_ _for_ _prores-ks_ **profile _**integer_ Select the ProRes profile to encode **proxy** **lt** **standard** **hq** **4444** **4444xq** **quant_mat _**integer_ Select quantization matrix. **auto** **default** **proxy** **lt** **standard** **hq** If set to _auto_, the matrix matching the profile will be picked. If not set, the matrix providing the highest quality, _default_, will be picked. **bits_per_mb _**integer_ How many bits to allot for coding one macroblock. Different profiles use between 200 and 2400 bits per macroblock, the maximum is 8000. **mbs_per_slice _**integer_ Number of macroblocks in each slice (1-8); the default value (8) should be good in almost all situations. **vendor _**string_ Override the 4-byte vendor ID. A custom vendor ID like _apl0_ would claim the stream was produced by the Apple encoder. **alpha_bits _**integer_ Specify number of bits for alpha component. Possible values are _0_, _8_ and _16_. Use _0_ to disable alpha plane coding. _Speed_ _considerations_ In the default mode of operation the encoder has to honor frame constraints (i.e. not produce frames with size bigger than requested) while still making output picture as good as possible. A frame containing a lot of small details is harder to compress and the encoder would spend more time searching for appropriate quantizers for each slice. Setting a higher **bits_per_mb **limit will improve the speed. For the fastest encoding speed set the **qscale **parameter (4 is the recommended value) and do not set a size constraint. ### QSV Encoders The family of Intel QuickSync Video encoders (MPEG-2, H.264, HEVC, JPEG/MJPEG, VP9, AV1) _Ratecontrol_ _Method_ The ratecontrol method is selected as follows: • When **global_quality **is specified, a quality-based mode is used. Specifically this means either - _CQP_ - constant quantizer scale, when the **qscale **codec flag is also set (the **-qscale** ffmpeg option). - _LA_ICQ_ - intelligent constant quality with lookahead, when the **look_ahead **option is also set. - _ICQ_ -- intelligent constant quality otherwise. For the ICQ modes, global quality range is 1 to 51, with 1 being the best quality. • Otherwise, a bitrate-based mode is used. For all of those, you should specify at least the desired average bitrate with the **b **option. - _LA_ - VBR with lookahead, when the **look_ahead **option is specified. - _VCM_ - video conferencing mode, when the **vcm **option is set. - _CBR_ - constant bitrate, when **maxrate **is specified and equal to the average bitrate. - _VBR_ - variable bitrate, when **maxrate **is specified, but is higher than the average bitrate. - _AVBR_ - average VBR mode, when **maxrate **is not specified, both **avbr_accuracy **and **avbr_convergence **are set to non-zero. This mode is available for H264 and HEVC on Windows. Note that depending on your system, a different mode than the one you specified may be selected by the encoder. Set the verbosity level to _verbose_ or higher to see the actual settings used by the QSV runtime. _Global_ _Options_ _->_ _MSDK_ _Options_ Additional libavcodec global options are mapped to MSDK options as follows: • **g/gop_size **-> **GopPicSize** • **bf/max_b_frames**+1 -> **GopRefDist** • **rc_init_occupancy/rc_initial_buffer_occupancy **-> **InitialDelayInKB** • **slices **-> **NumSlice** • **refs **-> **NumRefFrame** • **b_strategy/b_frame_strategy **-> **BRefType** • **cgop/CLOSED_GOP **codec flag -> **GopOptFlag** • For the _CQP_ mode, the **i_qfactor/i_qoffset **and **b_qfactor/b_qoffset **set the difference between _QPP_ and _QPI_, and _QPP_ and _QPB_ respectively. • Setting the **coder **option to the value _vlc_ will make the H.264 encoder use CAVLC instead of CABAC. _Common_ _Options_ Following options are used by all qsv encoders. _async_depth_ Specifies how many asynchronous operations an application performs before the application explicitly synchronizes the result. If zero, the value is not specified. _preset_ This option itemizes a range of choices from veryfast (best speed) to veryslow (best quality). **veryfast** **faster** **fast** **medium** **slow** **slower** **veryslow** _forced_idr_ Forcing I frames as IDR frames. _low_power_ For encoders set this flag to ON to reduce power consumption and GPU usage. _Runtime_ _Options_ Following options can be used durning qsv encoding. _global_quality_ _i_quant_factor_ _i_quant_offset_ _b_quant_factor_ _b_quant_offset_ Supported in h264_qsv and hevc_qsv. Change these value to reset qsv codec's qp configuration. _max_frame_size_ Supported in h264_qsv and hevc_qsv. Change this value to reset qsv codec's MaxFrameSize configuration. _gop_size_ Change this value to reset qsv codec's gop configuration. _int_ref_type_ _int_ref_cycle_size_ _int_ref_qp_delta_ _int_ref_cycle_dist_ Supported in h264_qsv and hevc_qsv. Change these value to reset qsv codec's Intra Refresh configuration. _qmax_ _qmin_ _max_qp_i_ _min_qp_i_ _max_qp_p_ _min_qp_p_ _max_qp_b_ _min_qp_b_ Supported in h264_qsv. Change these value to reset qsv codec's max/min qp configuration. _low_delay_brc_ Supported in h264_qsv, hevc_qsv and av1_qsv. Change this value to reset qsv codec's low_delay_brc configuration. _framerate_ Change this value to reset qsv codec's framerate configuration. _bit_rate_ _rc_buffer_size_ _rc_initial_buffer_occupancy_ _rc_max_rate_ Change these value to reset qsv codec's bitrate control configuration. _pic_timing_sei_ Supported in h264_qsv and hevc_qsv. Change this value to reset qsv codec's pic_timing_sei configuration. _H264_ _options_ These options are used by h264_qsv _extbrc_ Extended bitrate control. _recovery_point_sei_ Set this flag to insert the recovery point SEI message at the beginning of every intra refresh cycle. _rdo_ Enable rate distortion optimization. _max_frame_size_ Maximum encoded frame size in bytes. _max_frame_size_i_ Maximum encoded frame size for I frames in bytes. If this value is set as larger than zero, then for I frames the value set by max_frame_size is ignored. _max_frame_size_p_ Maximum encoded frame size for P frames in bytes. If this value is set as larger than zero, then for P frames the value set by max_frame_size is ignored. _max_slice_size_ Maximum encoded slice size in bytes. _bitrate_limit_ Toggle bitrate limitations. Modifies bitrate to be in the range imposed by the QSV encoder. Setting this flag off may lead to violation of HRD conformance. Mind that specifying bitrate below the QSV encoder range might significantly affect quality. If on this option takes effect in non CQP modes: if bitrate is not in the range imposed by the QSV encoder, it will be changed to be in the range. _mbbrc_ Setting this flag enables macroblock level bitrate control that generally improves subjective visual quality. Enabling this flag may have negative impact on performance and objective visual quality metric. _low_delay_brc_ Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides more accurate bitrate control to minimize the variance of bitstream size frame by frame. Value: -1-default 0-off 1-on _adaptive_i_ This flag controls insertion of I frames by the QSV encoder. Turn ON this flag to allow changing of frame type from P and B to I. _adaptive_b_ This flag controls changing of frame type from B to P. _p_strategy_ Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to 0). _b_strategy_ This option controls usage of B frames as reference. _dblk_idc_ This option disable deblocking. It has value in range 0~2. _cavlc_ If set, CAVLC is used; if unset, CABAC is used for encoding. _vcm_ Video conferencing mode, please see ratecontrol method. _idr_interval_ Distance (in I-frames) between IDR frames. _pic_timing_sei_ Insert picture timing SEI with pic_struct_syntax element. _single_sei_nal_unit_ Put all the SEI messages into one NALU. _max_dec_frame_buffering_ Maximum number of frames buffered in the DPB. _look_ahead_ Use VBR algorithm with look ahead. _look_ahead_depth_ Depth of look ahead in number frames. _look_ahead_downsampling_ Downscaling factor for the frames saved for the lookahead analysis. **unknown** **auto** **off** **2x** **4x** _int_ref_type_ Specifies intra refresh type. The major goal of intra refresh is improvement of error resilience without significant impact on encoded bitstream size caused by I frames. The SDK encoder achieves this by encoding part of each frame in refresh cycle using intra MBs. _none_ means no refresh. _vertical_ means vertical refresh, by column of MBs. _horizontal_ means horizontal refresh, by rows of MBs. _slice_ means horizontal refresh by slices without overlapping. In case of _slice_, in_ref_cycle_size is ignored. To enable intra refresh, B frame should be set to 0. _int_ref_cycle_size_ Specifies number of pictures within refresh cycle starting from 2. 0 and 1 are invalid values. _int_ref_qp_delta_ Specifies QP difference for inserted intra MBs. This is signed value in [-51, 51] range if target encoding bit-depth for luma samples is 8 and this range is [-63, 63] for 10 bit-depth or [-75, 75] for 12 bit-depth respectively. _int_ref_cycle_dist_ Distance between the beginnings of the intra-refresh cycles in frames. _profile_ **unknown** **baseline** **main** **high** _a53cc_ Use A53 Closed Captions (if available). _aud_ Insert the Access Unit Delimiter NAL. _mfmode_ Multi-Frame Mode. **off** **auto** _repeat_pps_ Repeat pps for every frame. _max_qp_i_ Maximum video quantizer scale for I frame. _min_qp_i_ Minimum video quantizer scale for I frame. _max_qp_p_ Maximum video quantizer scale for P frame. _min_qp_p_ Minimum video quantizer scale for P frame. _max_qp_b_ Maximum video quantizer scale for B frame. _min_qp_b_ Minimum video quantizer scale for B frame. _scenario_ Provides a hint to encoder about the scenario for the encoding session. **unknown** **displayremoting** **videoconference** **archive** **livestreaming** **cameracapture** **videosurveillance** **gamestreaming** **remotegaming** _avbr_accuracy_ Accuracy of the AVBR ratecontrol (unit of tenth of percent). _avbr_convergence_ Convergence of the AVBR ratecontrol (unit of 100 frames) The parameters _avbr_accuracy_ and _avbr_convergence_ are for the average variable bitrate control (AVBR) algorithm. The algorithm focuses on overall encoding quality while meeting the specified bitrate, _target_bitrate_, within the accuracy range _avbr_accuracy_, after a _avbr_Convergence_ period. This method does not follow HRD and the instant bitrate is not capped or padded. _skip_frame_ Use per-frame metadata "qsv_skip_frame" to skip frame when encoding. This option defines the usage of this metadata. **no_skip** Frame skipping is disabled. **insert_dummy** Encoder inserts into bitstream frame where all macroblocks are encoded as skipped. **insert_nothing** Similar to insert_dummy, but encoder inserts nothing into bitstream. The skipped frames are still used in brc. For example, gop still include skipped frames, and the frames after skipped frames will be larger in size. **brc_only** skip_frame metadata indicates the number of missed frames before the current frame. _HEVC_ _Options_ These options are used by hevc_qsv _extbrc_ Extended bitrate control. _recovery_point_sei_ Set this flag to insert the recovery point SEI message at the beginning of every intra refresh cycle. _rdo_ Enable rate distortion optimization. _max_frame_size_ Maximum encoded frame size in bytes. _max_frame_size_i_ Maximum encoded frame size for I frames in bytes. If this value is set as larger than zero, then for I frames the value set by max_frame_size is ignored. _max_frame_size_p_ Maximum encoded frame size for P frames in bytes. If this value is set as larger than zero, then for P frames the value set by max_frame_size is ignored. _max_slice_size_ Maximum encoded slice size in bytes. _mbbrc_ Setting this flag enables macroblock level bitrate control that generally improves subjective visual quality. Enabling this flag may have negative impact on performance and objective visual quality metric. _low_delay_brc_ Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides more accurate bitrate control to minimize the variance of bitstream size frame by frame. Value: -1-default 0-off 1-on _adaptive_i_ This flag controls insertion of I frames by the QSV encoder. Turn ON this flag to allow changing of frame type from P and B to I. _adaptive_b_ This flag controls changing of frame type from B to P. _p_strategy_ Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to 0). _b_strategy_ This option controls usage of B frames as reference. _dblk_idc_ This option disable deblocking. It has value in range 0~2. _idr_interval_ Distance (in I-frames) between IDR frames. **begin_only** Output an IDR-frame only at the beginning of the stream. _load_plugin_ A user plugin to load in an internal session. **none** **hevc_sw** **hevc_hw** _load_plugins_ A :-separate list of hexadecimal plugin UIDs to load in an internal session. _look_ahead_depth_ Depth of look ahead in number frames, available when extbrc option is enabled. _profile_ Set the encoding profile (scc requires libmfx >= 1.32). **unknown** **main** **main10** **mainsp** **rext** **scc** _tier_ Set the encoding tier (only level >= 4 can support high tier). This option only takes effect when the level option is specified. **main** **high** _gpb_ 1: GPB (generalized P/B frame) 0: regular P frame. _tile_cols_ Number of columns for tiled encoding. _tile_rows_ Number of rows for tiled encoding. _aud_ Insert the Access Unit Delimiter NAL. _pic_timing_sei_ Insert picture timing SEI with pic_struct_syntax element. _transform_skip_ Turn this option ON to enable transformskip. It is supported on platform equal or newer than ICL. _int_ref_type_ Specifies intra refresh type. The major goal of intra refresh is improvement of error resilience without significant impact on encoded bitstream size caused by I frames. The SDK encoder achieves this by encoding part of each frame in refresh cycle using intra MBs. _none_ means no refresh. _vertical_ means vertical refresh, by column of MBs. _horizontal_ means horizontal refresh, by rows of MBs. _slice_ means horizontal refresh by slices without overlapping. In case of _slice_, in_ref_cycle_size is ignored. To enable intra refresh, B frame should be set to 0. _int_ref_cycle_size_ Specifies number of pictures within refresh cycle starting from 2. 0 and 1 are invalid values. _int_ref_qp_delta_ Specifies QP difference for inserted intra MBs. This is signed value in [-51, 51] range if target encoding bit-depth for luma samples is 8 and this range is [-63, 63] for 10 bit-depth or [-75, 75] for 12 bit-depth respectively. _int_ref_cycle_dist_ Distance between the beginnings of the intra-refresh cycles in frames. _max_qp_i_ Maximum video quantizer scale for I frame. _min_qp_i_ Minimum video quantizer scale for I frame. _max_qp_p_ Maximum video quantizer scale for P frame. _min_qp_p_ Minimum video quantizer scale for P frame. _max_qp_b_ Maximum video quantizer scale for B frame. _min_qp_b_ Minimum video quantizer scale for B frame. _scenario_ Provides a hint to encoder about the scenario for the encoding session. **unknown** **displayremoting** **videoconference** **archive** **livestreaming** **cameracapture** **videosurveillance** **gamestreaming** **remotegaming** _avbr_accuracy_ Accuracy of the AVBR ratecontrol (unit of tenth of percent). _avbr_convergence_ Convergence of the AVBR ratecontrol (unit of 100 frames) The parameters _avbr_accuracy_ and _avbr_convergence_ are for the average variable bitrate control (AVBR) algorithm. The algorithm focuses on overall encoding quality while meeting the specified bitrate, _target_bitrate_, within the accuracy range _avbr_accuracy_, after a _avbr_Convergence_ period. This method does not follow HRD and the instant bitrate is not capped or padded. _skip_frame_ Use per-frame metadata "qsv_skip_frame" to skip frame when encoding. This option defines the usage of this metadata. **no_skip** Frame skipping is disabled. **insert_dummy** Encoder inserts into bitstream frame where all macroblocks are encoded as skipped. **insert_nothing** Similar to insert_dummy, but encoder inserts nothing into bitstream. The skipped frames are still used in brc. For example, gop still include skipped frames, and the frames after skipped frames will be larger in size. **brc_only** skip_frame metadata indicates the number of missed frames before the current frame. _MPEG2_ _Options_ These options are used by mpeg2_qsv _profile_ **unknown** **simple** **main** **high** _VP9_ _Options_ These options are used by vp9_qsv _profile_ **unknown** **profile0** **profile1** **profile2** **profile3** _tile_cols_ Number of columns for tiled encoding (requires libmfx >= 1.29). _tile_rows_ Number of rows for tiled encoding (requires libmfx >= 1.29). _AV1_ _Options_ These options are used by av1_qsv (requires libvpl). _profile_ **unknown** **main** _tile_cols_ Number of columns for tiled encoding. _tile_rows_ Number of rows for tiled encoding. _adaptive_i_ This flag controls insertion of I frames by the QSV encoder. Turn ON this flag to allow changing of frame type from P and B to I. _adaptive_b_ This flag controls changing of frame type from B to P. _b_strategy_ This option controls usage of B frames as reference. _extbrc_ Extended bitrate control. _look_ahead_depth_ Depth of look ahead in number frames, available when extbrc option is enabled. _low_delay_brc_ Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides more accurate bitrate control to minimize the variance of bitstream size frame by frame. Value: -1-default 0-off 1-on ### max_frame_size Set the allowed max size in bytes for each frame. If the frame size exceeds the limitation, encoder will adjust the QP value to control the frame size. Invalid in CQP rate control mode. ### snow _Options_ ### iterative_dia_size dia size for the iterative motion estimation ### VAAPI encoders Wrappers for hardware encoders accessible via VAAPI. These encoders only accept input in VAAPI hardware surfaces. If you have input in software frames, use the **hwupload **filter to upload them to the GPU. The following standard libavcodec options are used: • **g **/ **gop_size** • **bf **/ **max_b_frames** • **profile** If not set, this will be determined automatically from the format of the input frames and the profiles supported by the driver. • **level** • **b **/ **bit_rate** • **maxrate **/ **rc_max_rate** • **bufsize **/ **rc_buffer_size** • **rc_init_occupancy **/ **rc_initial_buffer_occupancy** • **compression_level** Speed / quality tradeoff: higher values are faster / worse quality. • **q **/ **global_quality** Size / quality tradeoff: higher values are smaller / worse quality. • **qmin** • **qmax** • **i_qfactor **/ **i_quant_factor** • **i_qoffset **/ **i_quant_offset** • **b_qfactor **/ **b_quant_factor** • **b_qoffset **/ **b_quant_offset** • **slices** All encoders support the following options: ### low_power Some drivers/platforms offer a second encoder for some codecs intended to use less power than the default encoder; setting this option will attempt to use that encoder. Note that it may support a reduced feature set, so some other options may not be available in this mode. ### idr_interval Set the number of normal intra frames between full-refresh (IDR) frames in open-GOP mode. The intra frames are still IRAPs, but will not include global headers and may have non- decodable leading pictures. ### b_depth Set the B-frame reference depth. When set to one (the default), all B-frames will refer only to P- or I-frames. When set to greater values multiple layers of B-frames will be present, frames in each layer only referring to frames in higher layers. ### async_depth Maximum processing parallelism. Increase this to improve single channel performance. This option doesn't work if driver doesn't implement vaSyncBuffer function. Please make sure there are enough hw_frames allocated if a large number of async_depth is used. ### max_frame_size Set the allowed max size in bytes for each frame. If the frame size exceeds the limitation, encoder will adjust the QP value to control the frame size. Invalid in CQP rate control mode. ### rc_mode Set the rate control mode to use. A given driver may only support a subset of modes. Possible modes: **auto** Choose the mode automatically based on driver support and the other options. This is the default. **CQP **Constant-quality. **CBR **Constant-bitrate. **VBR **Variable-bitrate. **ICQ **Intelligent constant-quality. **QVBR** Quality-defined variable-bitrate. **AVBR** Average variable bitrate. Each encoder also has its own specific options: ### av1_vaapi **profile **sets the value of _seq_profile_. **tier **sets the value of _seq_tier_. **level **sets the value of _seq_level_idx_. **tiles** Set the number of tiles to encode the input video with, as columns x rows. (default is auto, which means use minimal tile column/row number). **tile_groups** Set tile groups number. All the tiles will be distributed as evenly as possible to each tile group. (default is 1). ### h264_vaapi **profile **sets the value of _profile_idc_ and the _constraint_set*_flag_s. **level **sets the value of _level_idc_. **coder** Set entropy encoder (default is _cabac_). Possible values: **ac** **cabac** Use CABAC. **vlc** **cavlc** Use CAVLC. **aud **Include access unit delimiters in the stream (not included by default). **sei **Set SEI message types to include. Some combination of the following values: **identifier** Include a _user_data_unregistered_ message containing information about the encoder. **timing** Include picture timing parameters (_buffering_period_ and _pic_timing_ messages). **recovery_point** Include recovery points where appropriate (_recovery_point_ messages). ### hevc_vaapi **profile **and **level **set the values of _general_profile_idc_ and _general_level_idc_ respectively. **aud **Include access unit delimiters in the stream (not included by default). **tier** Set _general_tier_flag_. This may affect the level chosen for the stream if it is not explicitly specified. **sei **Set SEI message types to include. Some combination of the following values: **hdr **Include HDR metadata if the input frames have it (_mastering_display_colour_volume_ and _content_light_level_ messages). **tiles** Set the number of tiles to encode the input video with, as columns x rows. Larger numbers allow greater parallelism in both encoding and decoding, but may decrease coding efficiency. ### mjpeg_vaapi Only baseline DCT encoding is supported. The encoder always uses the standard quantisation and huffman tables - **global_quality **scales the standard quantisation table (range 1-100). For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported. RGB is also supported, and will create an RGB JPEG. **jfif** Include JFIF header in each frame (not included by default). **huffman** Include standard huffman tables (on by default). Turning this off will save a few hundred bytes in each output frame, but may lose compatibility with some JPEG decoders which don't fully handle MJPEG. ### mpeg2_vaapi **profile **and **level **set the value of _profile_and_level_indication_. ### vp8_vaapi B-frames are not supported. **global_quality **sets the _q_idx_ used for non-key frames (range 0-127). **loop_filter_level** **loop_filter_sharpness** Manually set the loop filter parameters. ### vp9_vaapi **global_quality **sets the _q_idx_ used for P-frames (range 0-255). **loop_filter_level** **loop_filter_sharpness** Manually set the loop filter parameters. B-frames are supported, but the output stream is always in encode order rather than display order. If B-frames are enabled, it may be necessary to use the **vp9_raw_reorder** bitstream filter to modify the output stream to display frames in the correct order. Only normal frames are produced - the **vp9_superframe **bitstream filter may be required to produce a stream usable with all decoders. ### vbn Vizrt Binary Image encoder. This format is used by the broadcast vendor Vizrt for quick texture streaming. Advanced features of the format such as LZW compression of texture data or generation of mipmaps are not supported. _Options_ **format _**string_ Sets the texture compression used by the VBN file. Can be _dxt1_, _dxt5_ or _raw_. Default is _dxt5_. ### vc2 SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at professional broadcasting but since it supports yuv420, yuv422 and yuv444 at 8 (limited range or full range), 10 or 12 bits, this makes it suitable for other tasks which require low overhead and low compression (like screen recording). _Options_ **b **Sets target video bitrate. Usually that's around 1:6 of the uncompressed video bitrate (e.g. for 1920x1080 50fps yuv422p10 that's around 400Mbps). Higher values (close to the uncompressed bitrate) turn on lossless compression mode. ### field_order Enables field coding when set (e.g. to tt - top field first) for interlaced inputs. Should increase compression with interlaced content as it splits the fields and encodes each separately. ### wavelet_depth Sets the total amount of wavelet transforms to apply, between 1 and 5 (default). Lower values reduce compression and quality. Less capable decoders may not be able to handle values of **wavelet_depth **over 3. ### wavelet_type Sets the transform type. Currently only _5_3_ (LeGall) and _9_7_ (Deslauriers-Dubuc) are implemented, with 9_7 being the one with better compression and thus is the default. ### slice_width ### slice_height Sets the slice size for each slice. Larger values result in better compression. For compatibility with other more limited decoders use **slice_width **of 32 and **slice_height **of 8. ### tolerance Sets the undershoot tolerance of the rate control system in percent. This is to prevent an expensive search from being run. **qm **Sets the quantization matrix preset to use by default or when **wavelet_depth **is set to 5 - _default_ Uses the default quantization matrix from the specifications, extended with values for the fifth level. This provides a good balance between keeping detail and omitting artifacts. - _flat_ Use a completely zeroed out quantization matrix. This increases PSNR but might reduce perception. Use in bogus benchmarks. - _color_ Reduces detail but attempts to preserve color at extremely low bitrates. ## SUBTITLES ENCODERS ### dvdsub This codec encodes the bitmap subtitle format that is used in DVDs. Typically they are stored in VOBSUB file pairs (*.idx + *.sub), and they can also be used in Matroska files. _Options_ ### palette Specify the global palette used by the bitmaps. The format for this option is a string containing 16 24-bits hexadecimal numbers (without 0x prefix) separated by commas, for example "0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b". ### even_rows_fix When set to 1, enable a work-around that makes the number of pixel rows even in all subtitles. This fixes a problem with some players that cut off the bottom row if the number is odd. The work-around just adds a fully transparent row if needed. The overhead is low, typically one byte per subtitle on average. By default, this work-around is disabled. ## BITSTREAM FILTERS When you configure your FFmpeg build, all the supported bitstream filters are enabled by default. You can list all available ones using the configure option "--list-bsfs". You can disable all the bitstream filters using the configure option "--disable-bsfs", and selectively enable any bitstream filter using the option "--enable-bsf=BSF", or you can disable a particular bitstream filter using the option "--disable-bsf=BSF". The option "-bsfs" of the ff* tools will display the list of all the supported bitstream filters included in your build. The ff* tools have a -bsf option applied per stream, taking a comma-separated list of filters, whose parameters follow the filter name after a '='. ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT Below is a description of the currently available bitstream filters, with their parameters, if any. ### aac_adtstoasc Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration bitstream. This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS header and removes the ADTS header. This filter is required for example when copying an AAC stream from a raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or to MOV/MP4 files and related formats such as 3GP or M4A. Please note that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats. ### av1_metadata Modify metadata embedded in an AV1 stream. **td **Insert or remove temporal delimiter OBUs in all temporal units of the stream. **insert** Insert a TD at the beginning of every TU which does not already have one. **remove** Remove the TD from the beginning of every TU which has one. ### color_primaries ### transfer_characteristics ### matrix_coefficients Set the color description fields in the stream (see AV1 section 6.4.2). ### color_range Set the color range in the stream (see AV1 section 6.4.2; note that this cannot be set for streams using BT.709 primaries, sRGB transfer characteristic and identity (RGB) matrix coefficients). **tv **Limited range. **pc **Full range. ### chroma_sample_position Set the chroma sample location in the stream (see AV1 section 6.4.2). This can only be set for 4:2:0 streams. **vertical** Left position (matching the default in MPEG-2 and H.264). **colocated** Top-left position. ### tick_rate Set the tick rate (_time_scale_ _/_ _num_units_in_display_tick_) in the timing info in the sequence header. ### num_ticks_per_picture Set the number of ticks in each picture, to indicate that the stream has a fixed framerate. Ignored if **tick_rate **is not also set. ### delete_padding Deletes Padding OBUs. ### chomp Remove zero padding at the end of a packet. ### dca_core Extract the core from a DCA/DTS stream, dropping extensions such as DTS-HD. ### dump_extra Add extradata to the beginning of the filtered packets except when said packets already exactly begin with the extradata that is intended to be added. ### freq The additional argument specifies which packets should be filtered. It accepts the values: **k** **keyframe** add extradata to all key packets **e** **all **add extradata to all packets If not specified it is assumed **k**. For example the following **ffmpeg **command forces a global header (thus disabling individual packet headers) in the H.264 packets generated by the "libx264" encoder, but corrects them by adding the header stored in extradata to the key packets: ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts ### dv_error_marker Blocks in DV which are marked as damaged are replaced by blocks of the specified color. ### color The color to replace damaged blocks by **sta **A 16 bit mask which specifies which of the 16 possible error status values are to be replaced by colored blocks. 0xFFFE is the default which replaces all non 0 error status values. **ok **No error, no concealment **err **Error, No concealment **res **Reserved **notok** Error or concealment **notres** Not reserved **Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru** The specific error status code see page 44-46 or section 5.5 of <****> ### eac3_core Extract the core from a E-AC-3 stream, dropping extra channels. ### extract_extradata Extract the in-band extradata. Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers, or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-band" (i.e. as a part of the bitstream containing the coded frames) or "out of band" (e.g. on the container level). This latter form is called "extradata" in FFmpeg terminology. This bitstream filter detects the in-band headers and makes them available as extradata. ### remove When this option is enabled, the long-term headers are removed from the bitstream after extraction. ### filter_units Remove units with types in or not in a given set from the stream. ### pass_types List of unit types or ranges of unit types to pass through while removing all others. This is specified as a '|'-separated list of unit type values or ranges of values with '-'. ### remove_types Identical to **pass_types**, except the units in the given set removed and all others passed through. Extradata is unchanged by this transformation, but note that if the stream contains inline parameter sets then the output may be unusable if they are removed. For example, to remove all non-VCL NAL units from an H.264 stream: ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT To remove all AUDs, SEI and filler from an H.265 stream: ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT ### hapqa_extract Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file. ### texture Specifies the texture to keep. **color** **alpha** Convert HAPQA to HAPQ ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov Convert HAPQA to HAPAlphaOnly ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov ### h264_metadata Modify metadata embedded in an H.264 stream. **aud **Insert or remove AUD NAL units in all access units of the stream. **pass** **insert** **remove** Default is pass. ### sample_aspect_ratio Set the sample aspect ratio of the stream in the VUI parameters. See H.264 table E-1. ### overscan_appropriate_flag Set whether the stream is suitable for display using overscan or not (see H.264 section E.2.1). ### video_format ### video_full_range_flag Set the video format in the stream (see H.264 section E.2.1 and table E-2). ### colour_primaries ### transfer_characteristics ### matrix_coefficients Set the colour description in the stream (see H.264 section E.2.1 and tables E-3, E-4 and E-5). ### chroma_sample_loc_type Set the chroma sample location in the stream (see H.264 section E.2.1 and figure E-1). ### tick_rate Set the tick rate (time_scale / num_units_in_tick) in the VUI parameters. This is the smallest time unit representable in the stream, and in many cases represents the field rate of the stream (double the frame rate). ### fixed_frame_rate_flag Set whether the stream has fixed framerate - typically this indicates that the framerate is exactly half the tick rate, but the exact meaning is dependent on interlacing and the picture structure (see H.264 section E.2.1 and table E-6). ### zero_new_constraint_set_flags Zero constraint_set4_flag and constraint_set5_flag in the SPS. These bits were reserved in a previous version of the H.264 spec, and thus some hardware decoders require these to be zero. The result of zeroing this is still a valid bitstream. ### crop_left ### crop_right ### crop_top ### crop_bottom Set the frame cropping offsets in the SPS. These values will replace the current ones if the stream is already cropped. These fields are set in pixels. Note that some sizes may not be representable if the chroma is subsampled or the stream is interlaced (see H.264 section 7.4.2.1.1). ### sei_user_data Insert a string as SEI unregistered user data. The argument must be of the form _UUID+string_, where the UUID is as hex digits possibly separated by hyphens, and the string can be anything. For example, **086f3693-b7b3-4f2c-9653-21492feee5b8+hello **will insert the string ``hello'' associated with the given UUID. ### delete_filler Deletes both filler NAL units and filler SEI messages. ### display_orientation Insert, extract or remove Display orientation SEI messages. See H.264 section D.1.27 and D.2.27 for syntax and semantics. **pass** **insert** **remove** **extract** Default is pass. Insert mode works in conjunction with "rotate" and "flip" options. Any pre-existing Display orientation messages will be removed in insert or remove mode. Extract mode attaches the display matrix to the packet as side data. ### rotate Set rotation in display orientation SEI (anticlockwise angle in degrees). Range is -360 to +360. Default is NaN. ### flip Set flip in display orientation SEI. **horizontal** **vertical** Default is unset. ### level Set the level in the SPS. Refer to H.264 section A.3 and tables A-1 to A-5. The argument must be the name of a level (for example, **4.2**), a level_idc value (for example, **42**), or the special name **auto **indicating that the filter should attempt to guess the level from the input stream properties. ### h264_mp4toannexb Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.264 specification). This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer "mpegts"). For example to remux an MP4 file containing an H.264 stream to mpegts format with **ffmpeg**, you can use the command: ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts Please note that this filter is auto-inserted for MPEG-TS (muxer "mpegts") and raw H.264 (muxer "h264") output formats. ### h264_redundant_pps This applies a specific fixup to some Blu-ray streams which contain redundant PPSs modifying irrelevant parameters of the stream which confuse other transformations which require correct extradata. ### hevc_metadata Modify metadata embedded in an HEVC stream. **aud **Insert or remove AUD NAL units in all access units of the stream. **insert** **remove** ### sample_aspect_ratio Set the sample aspect ratio in the stream in the VUI parameters. ### video_format ### video_full_range_flag Set the video format in the stream (see H.265 section E.3.1 and table E.2). ### colour_primaries ### transfer_characteristics ### matrix_coefficients Set the colour description in the stream (see H.265 section E.3.1 and tables E.3, E.4 and E.5). ### chroma_sample_loc_type Set the chroma sample location in the stream (see H.265 section E.3.1 and figure E.1). ### tick_rate Set the tick rate in the VPS and VUI parameters (time_scale / num_units_in_tick). Combined with **num_ticks_poc_diff_one**, this can set a constant framerate in the stream. Note that it is likely to be overridden by container parameters when the stream is in a container. ### num_ticks_poc_diff_one Set poc_proportional_to_timing_flag in VPS and VUI and use this value to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and E.3.1). Ignored if **tick_rate **is not also set. ### crop_left ### crop_right ### crop_top ### crop_bottom Set the conformance window cropping offsets in the SPS. These values will replace the current ones if the stream is already cropped. These fields are set in pixels. Note that some sizes may not be representable if the chroma is subsampled (H.265 section 7.4.3.2.1). ### level Set the level in the VPS and SPS. See H.265 section A.4 and tables A.6 and A.7. The argument must be the name of a level (for example, **5.1**), a _general_level_idc_ value (for example, **153 **for level 5.1), or the special name **auto **indicating that the filter should attempt to guess the level from the input stream properties. ### hevc_mp4toannexb Convert an HEVC/H.265 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.265 specification). This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer "mpegts"). For example to remux an MP4 file containing an HEVC stream to mpegts format with **ffmpeg**, you can use the command: ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts Please note that this filter is auto-inserted for MPEG-TS (muxer "mpegts") and raw HEVC/H.265 (muxer "h265" or "hevc") output formats. ### imxdump Modifies the bitstream to fit in MOV and to be usable by the Final Cut Pro decoder. This filter only applies to the mpeg2video codec, and is likely not needed for Final Cut Pro 7 and newer with the appropriate **-tag:v**. For example, to remux 30 MB/sec NTSC IMX to MOV: ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov ### mjpeg2jpeg Convert MJPEG/AVI1 packets to full JPEG/JFIF packets. MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be extracted without loss, e.g. by ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for decoding. Quoting from <****>: Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that "MJPEG, or at least the MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* -- Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won't have any idea how to decompress the data. The exact table necessary is given in the OpenDML spec." This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images. ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg exiftran -i -9 frame*.jpg ffmpeg -i frame_%d.jpg -c:v copy rotated.avi ### mjpegadump Add an MJPEG A header to the bitstream, to enable decoding by Quicktime. ### mov2textsub Extract a representable text file from MOV subtitles, stripping the metadata header from each subtitle packet. See also the **text2movsub **filter. ### mp3decomp Decompress non-standard compressed MP3 audio headers. ### mpeg2_metadata Modify metadata embedded in an MPEG-2 stream. ### display_aspect_ratio Set the display aspect ratio in the stream. The following fixed values are supported: **4/3** **16/9** **221/100** Any other value will result in square pixels being signalled instead (see H.262 section 6.3.3 and table 6-3). ### frame_rate Set the frame rate in the stream. This is constructed from a table of known values combined with a small multiplier and divisor - if the supplied value is not exactly representable, the nearest representable value will be used instead (see H.262 section 6.3.3 and table 6-4). ### video_format Set the video format in the stream (see H.262 section 6.3.6 and table 6-6). ### colour_primaries ### transfer_characteristics ### matrix_coefficients Set the colour description in the stream (see H.262 section 6.3.6 and tables 6-7, 6-8 and 6-9). ### mpeg4_unpack_bframes Unpack DivX-style packed B-frames. DivX-style packed B-frames are not valid MPEG-4 and were only a workaround for the broken Video for Windows subsystem. They use more space, can cause minor AV sync issues, require more CPU power to decode (unless the player has some decoded picture queue to compensate the 2,0,2,0 frame per packet style) and cause trouble if copied into a standard container like mp4 or mpeg-ps/ts, because MPEG-4 decoders may not be able to decode them, since they are not valid MPEG-4. For example to fix an AVI file containing an MPEG-4 stream with DivX-style packed B-frames using **ffmpeg**, you can use the command: ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi ### noise Damages the contents of packets or simply drops them without damaging the container. Can be used for fuzzing or testing error resilience/concealment. Parameters: ### amount Accepts an expression whose evaluation per-packet determines how often bytes in that packet will be modified. A value below 0 will result in a variable frequency. Default is 0 which results in no modification. However, if neither amount nor drop is specified, amount will be set to _-1_. See below for accepted variables. ### drop Accepts an expression evaluated per-packet whose value determines whether that packet is dropped. Evaluation to a positive value results in the packet being dropped. Evaluation to a negative value results in a variable chance of it being dropped, roughly inverse in proportion to the magnitude of the value. Default is 0 which results in no drops. See below for accepted variables. ### dropamount Accepts a non-negative integer, which assigns a variable chance of it being dropped, roughly inverse in proportion to the value. Default is 0 which results in no drops. This option is kept for backwards compatibility and is equivalent to setting drop to a negative value with the same magnitude i.e. "dropamount=4" is the same as "drop=-4". Ignored if drop is also specified. Both "amount" and "drop" accept expressions containing the following variables: **n **The index of the packet, starting from zero. **tb **The timebase for packet timestamps. **pts **Packet presentation timestamp. **dts **Packet decoding timestamp. ### nopts Constant representing AV_NOPTS_VALUE. ### startpts First non-AV_NOPTS_VALUE PTS seen in the stream. ### startdts First non-AV_NOPTS_VALUE DTS seen in the stream. ### duration **d **Packet duration, in timebase units. **pos **Packet position in input; may be -1 when unknown or not set. ### size Packet size, in bytes. **key **Whether packet is marked as a keyframe. ### state A pseudo random integer, primarily derived from the content of packet payload. _Examples_ Apply modification to every byte but don't drop any packets. ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv Drop every video packet not marked as a keyframe after timestamp 30s but do not modify any of the remaining packets. ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv Drop one second of audio every 10 seconds and add some random noise to the rest. ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv ### null This bitstream filter passes the packets through unchanged. ### pcm_rechunk Repacketize PCM audio to a fixed number of samples per packet or a fixed packet rate per second. This is similar to the **asetnsamples audio filter **but works on audio packets instead of audio frames. ### nb_out_samples, n Set the number of samples per each output audio packet. The number is intended as the number of samples _per_ _each_ _channel_. Default value is 1024. ### pad, p If set to 1, the filter will pad the last audio packet with silence, so that it will contain the same number of samples (or roughly the same number of samples, see **frame_rate**) as the previous ones. Default value is 1. ### frame_rate, r This option makes the filter output a fixed number of packets per second instead of a fixed number of samples per packet. If the audio sample rate is not divisible by the frame rate then the number of samples will not be constant but will vary slightly so that each packet will start as close to the frame boundary as possible. Using this option has precedence over **nb_out_samples**. You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio for NTSC frame rate using the **frame_rate **option. ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc - ### pgs_frame_merge Merge a sequence of PGS Subtitle segments ending with an "end of display set" segment into a single packet. This is required by some containers that support PGS subtitles (muxer "matroska"). ### prores_metadata Modify color property metadata embedded in prores stream. ### color_primaries Set the color primaries. Available values are: **auto** Keep the same color primaries property (default). **unknown** **bt709** **bt470bg** BT601 625 **smpte170m** BT601 525 **bt2020** **smpte431** DCI P3 **smpte432** P3 D65 ### transfer_characteristics Set the color transfer. Available values are: **auto** Keep the same transfer characteristics property (default). **unknown** **bt709** BT 601, BT 709, BT 2020 **smpte2084** SMPTE ST 2084 **arib-std-b67** ARIB STD-B67 ### matrix_coefficients Set the matrix coefficient. Available values are: **auto** Keep the same colorspace property (default). **unknown** **bt709** **smpte170m** BT 601 **bt2020nc** Set Rec709 colorspace for each frame of the file ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov Set Hybrid Log-Gamma parameters for each frame of the file ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov ### remove_extra Remove extradata from packets. It accepts the following parameter: ### freq Set which frame types to remove extradata from. **k **Remove extradata from non-keyframes only. **keyframe** Remove extradata from keyframes only. **e, all** Remove extradata from all frames. ### setts Set PTS and DTS in packets. It accepts the following parameters: **ts** ### pts **dts **Set expressions for PTS, DTS or both. ### duration Set expression for duration. ### time_base Set output time base. The expressions are evaluated through the eval API and can contain the following constants: **N **The count of the input packet. Starting from 0. **TS **The demux timestamp in input in case of "ts" or "dts" option or presentation timestamp in case of "pts" option. **POS **The original position in the file of the packet, or undefined if undefined for the current packet **DTS **The demux timestamp in input. **PTS **The presentation timestamp in input. **DURATION** The duration in input. **STARTDTS** The DTS of the first packet. **STARTPTS** The PTS of the first packet. **PREV_INDTS** The previous input DTS. **PREV_INPTS** The previous input PTS. **PREV_INDURATION** The previous input duration. **PREV_OUTDTS** The previous output DTS. **PREV_OUTPTS** The previous output PTS. **PREV_OUTDURATION** The previous output duration. **NEXT_DTS** The next input DTS. **NEXT_PTS** The next input PTS. **NEXT_DURATION** The next input duration. **TB **The timebase of stream packet belongs. **TB_OUT** The output timebase. **SR **The sample rate of stream packet belongs. **NOPTS** The AV_NOPTS_VALUE constant. ### text2movsub Convert text subtitles to MOV subtitles (as used by the "mov_text" codec) with metadata headers. See also the **mov2textsub **filter. ### trace_headers Log trace output containing all syntax elements in the coded stream headers (everything above the level of individual coded blocks). This can be useful for debugging low-level stream issues. Supports AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending on the build only a subset of these may be available. ### truehd_core Extract the core from a TrueHD stream, dropping ATMOS data. ### vp9_metadata Modify metadata embedded in a VP9 stream. ### color_space Set the color space value in the frame header. Note that any frame set to RGB will be implicitly set to PC range and that RGB is incompatible with profiles 0 and 2. **unknown** **bt601** **bt709** **smpte170** **smpte240** **bt2020** **rgb** ### color_range Set the color range value in the frame header. Note that any value imposed by the color space will take precedence over this value. **tv** **pc** ### vp9_superframe Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This fixes merging of split/segmented VP9 streams where the alt-ref frame was split from its visible counterpart. ### vp9_superframe_split Split VP9 superframes into single frames. ### vp9_raw_reorder Given a VP9 stream with correct timestamps but possibly out of order, insert additional show- existing-frame packets to correct the ordering. ## FORMAT OPTIONS The libavformat library provides some generic global options, which can be set on all the muxers and demuxers. In addition each muxer or demuxer may support so-called private options, which are specific for that component. Options may be set by specifying -_option_ _value_ in the FFmpeg tools, or by setting the value explicitly in the "AVFormatContext" options or using the _libavutil/opt.h_ API for programmatic use. The list of supported options follows: **avioflags _**flags_ **(_**input/output_**)** Possible values: **direct** Reduce buffering. **probesize _**integer_ **(_**input_**)** Set probing size in bytes, i.e. the size of the data to analyze to get stream information. A higher value will enable detecting more information in case it is dispersed into the stream, but will increase latency. Must be an integer not lesser than 32. It is 5000000 by default. **max_probe_packets _**integer_ **(_**input_**)** Set the maximum number of buffered packets when probing a codec. Default is 2500 packets. **packetsize _**integer_ **(_**output_**)** Set packet size. **fflags _**flags_ Set format flags. Some are implemented for a limited number of formats. Possible values for input files: **discardcorrupt** Discard corrupted packets. **fastseek** Enable fast, but inaccurate seeks for some formats. **genpts** Generate missing PTS if DTS is present. **igndts** Ignore DTS if PTS is set. Inert when nofillin is set. **ignidx** Ignore index. **nobuffer** Reduce the latency introduced by buffering during initial input streams analysis. **nofillin** Do not fill in missing values in packet fields that can be exactly calculated. **noparse** Disable AVParsers, this needs "+nofillin" too. **sortdts** Try to interleave output packets by DTS. At present, available only for AVIs with an index. Possible values for output files: **autobsf** Automatically apply bitstream filters as required by the output format. Enabled by default. **bitexact** Only write platform-, build- and time-independent data. This ensures that file and data checksums are reproducible and match between platforms. Its primary use is for regression testing. **flush_packets** Write out packets immediately. **shortest** Stop muxing at the end of the shortest stream. It may be needed to increase max_interleave_delta to avoid flushing the longer streams before EOF. **seek2any _**integer_ **(_**input_**)** Allow seeking to non-keyframes on demuxer level when supported if set to 1. Default is 0. **analyzeduration _**integer_ **(_**input_**)** Specify how many microseconds are analyzed to probe the input. A higher value will enable detecting more accurate information, but will increase latency. It defaults to 5,000,000 microseconds = 5 seconds. **cryptokey _**hexadecimal_ _string_ **(_**input_**)** Set decryption key. **indexmem _**integer_ **(_**input_**)** Set max memory used for timestamp index (per stream). **rtbufsize _**integer_ **(_**input_**)** Set max memory used for buffering real-time frames. **fdebug _**flags_ **(_**input/output_**)** Print specific debug info. Possible values: **ts** **max_delay _**integer_ **(_**input/output_**)** Set maximum muxing or demuxing delay in microseconds. **fpsprobesize _**integer_ **(_**input_**)** Set number of frames used to probe fps. **audio_preload _**integer_ **(_**output_**)** Set microseconds by which audio packets should be interleaved earlier. **chunk_duration _**integer_ **(_**output_**)** Set microseconds for each chunk. **chunk_size _**integer_ **(_**output_**)** Set size in bytes for each chunk. **err_detect, f_err_detect _**flags_ **(_**input_**)** Set error detection flags. "f_err_detect" is deprecated and should be used only via the **ffmpeg **tool. Possible values: **crccheck** Verify embedded CRCs. **bitstream** Detect bitstream specification deviations. **buffer** Detect improper bitstream length. **explode** Abort decoding on minor error detection. **careful** Consider things that violate the spec and have not been seen in the wild as errors. **compliant** Consider all spec non compliancies as errors. **aggressive** Consider things that a sane encoder should not do as an error. **max_interleave_delta _**integer_ **(_**output_**)** Set maximum buffering duration for interleaving. The duration is expressed in microseconds, and defaults to 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 _**integer_ **(_**input_**)** Use wallclock as timestamps if set to 1. Default is 0. **avoid_negative_ts _**integer_ **(_**output_**)** Possible values: **make_non_negative** Shift timestamps to make them non-negative. Also note that this affects only leading negative timestamps, and not non-monotonic negative timestamps. **make_zero** Shift timestamps so that the first timestamp is 0. **auto (default)** Enables shifting when required by the target format. **disabled** Disables shifting of timestamp. When shifting is enabled, all output timestamps are shifted by the same amount. Audio, video, and subtitles desynching and relative timestamp differences are preserved compared to how they would have been without shifting. **skip_initial_bytes _**integer_ **(_**input_**)** Set number of bytes to skip before reading header and frames if set to 1. Default is 0. **correct_ts_overflow _**integer_ **(_**input_**)** Correct single timestamp overflows if set to 1. Default is 1. **flush_packets _**integer_ **(_**output_**)** Flush the underlying I/O stream after each packet. Default 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. **output_ts_offset _**offset_ **(_**output_**)** Set the output time offset. _offset_ must be a time duration specification, see **the Time duration section in the** [**ffmpeg-utils(1)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown) manual**. The offset is added by the muxer to the output timestamps. Specifying a positive offset means that the corresponding streams are delayed bt the time duration specified in _offset_. Default value is 0 (meaning that no offset is applied). **format_whitelist _**list_ **(_**input_**)** "," separated list of allowed demuxers. By default all are allowed. **dump_separator _**string_ **(_**input_**)** Separator used to separate the fields printed on the command line about the Stream parameters. For example, to separate the fields with newlines and indentation: ffprobe -dump_separator " " -i ~/videos/matrixbench_mpeg2.mpg **max_streams _**integer_ **(_**input_**)** 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_estimate_duration_from_pts _**bool_ **(_**input_**)** Skip estimation of input duration when calculated using PTS. At present, applicable for MPEG-PS and MPEG-TS. **strict, f_strict _**integer_ **(_**input/output_**)** Specify how strictly to follow the standards. "f_strict" is deprecated and should be used only via the **ffmpeg **tool. Possible values: **very** strictly conform to an older more strict version of the spec or reference software **strict** strictly conform to all the things in the spec no matter what consequences **normal** **unofficial** allow unofficial extensions **experimental** allow non standardized experimental things, experimental (unfinished/work in progress/not well tested) decoders and encoders. Note: experimental decoders can pose a security risk, do not use this for decoding untrusted input. ### Format stream specifiers 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)](https://www.chedong.com/phpMan.php/man/ffmpeg/1/markdown) manual**. ## DEMUXERS Demuxers are configured elements in FFmpeg that can read the multimedia streams from a particular type of file. When you configure your FFmpeg build, all the supported demuxers are enabled by default. You can list all available ones using the configure option "--list-demuxers". You can disable all the demuxers using the configure option "--disable-demuxers", and selectively enable a single demuxer with the option "--enable-demuxer=_DEMUXER_", or disable it with the option "--disable-demuxer=_DEMUXER_". The option "-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. **aa** Audible Format 2, 3, and 4 demuxer. This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files. ### aac 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. ### apng Animated Portable Network Graphics demuxer. This demuxer is used to demux APNG files. All headers, but the PNG signature, up to (but not including) the first fcTL chunk are transmitted as extradata. Frames are then split as being all the chunks between two fcTL ones, or between the last fcTL and IEND chunks. ### -ignore_loop _ Ignore the loop variable in the file if set. Default is enabled. ### -max_fps _ Maximum framerate in frames per second. Default of 0 imposes no limit. ### -default_fps _ Default framerate in frames per second when none is specified in the file (0 meaning as fast as possible). Default is 15. ### asf Advanced Systems Format demuxer. This demuxer is used to demux ASF files and MMS network streams. ### -no_resync_search _ Do not try to resynchronize by looking for a certain optional start code. ### concat Virtual concatenation script demuxer. This demuxer reads a list of files and other directives from a text file and demuxes them one after the other, as if all their packets had been muxed together. The timestamps in the files are adjusted so that the first file starts at 0 and each next file starts where the previous one finishes. Note that it is done globally and may cause gaps if all streams do not have exactly the same length. All files must have the same streams (same codecs, same time base, etc.). The duration of each file is used to adjust the timestamps of the next file: if the duration is incorrect (because it was computed using the bit-rate or because the file is truncated, for example), it can cause artifacts. The "duration" directive can be used to override the duration stored in each file. _Syntax_ The script is a text file in extended-ASCII, with one directive per line. Empty lines, leading spaces and lines starting with '#' are ignored. The following directive is recognized: ### "file path" Path to a file to read; special characters and spaces must be escaped with backslash or single quotes. All subsequent file-related directives apply to that file. ### "ffconcat version 1.0" Identify the script type and version. 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: ### safe 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. ### auto_convert If set to 1, try to perform automatic conversions on packet data to make the streams concatenable. The default is 1. Currently, the only conversion is adding the h264_mp4toannexb bitstream filter to H.264 streams in MP4 format. This is necessary in particular if there are resolution changes. ### segment_time_metadata If set to 1, every packet will contain the _lavf.concat.start_time_ and the _lavf.concat.duration_ packet metadata values which are the start_time and the duration of the respective file segments in the concatenated output expressed in microseconds. The duration metadata is only set if it is known based on the concat file. The default is 0. _Examples_ • Use absolute filenames and include some comments: # my first filename file /mnt/share/file-1.wav # my second filename including whitespace file '/mnt/share/file 2.wav' # my third filename including whitespace plus single quote file '/mnt/share/file 3'\''.wav' • Allow for input format auto-probing, use safe filenames and set the duration of the first file: ffconcat version 1.0 file file-1.wav duration 20.0 file subdir/file-2.wav ### dash 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 "" as metadata keys named "id" and "variant_bitrate" respectively. _Options_ This demuxer accepts the following option: ### cenc_decryption_key 16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7). **ea** Electronic Arts Multimedia format demuxer. This format is used by various Electronic Arts games. _Options_ **merge_alpha _**bool_ 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. ### imf Interoperable Master Format demuxer. This demuxer presents audio and video streams found in an IMF Composition, as specified in <****>. ffmpeg [-assetmaps ,,...] -i ... If "-assetmaps" is not specified, the demuxer looks for a file called _ASSETMAP.xml_ in the same directory as the CPL. ### flv, live_flv, kux 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:///anything/key .... ### -flv_metadata _ Allocate the streams according to the onMetaData array content. ### -flv_ignore_prevtag _ Ignore the size of previous tag value. ### -flv_full_metadata _ Output all context of the onMetadata. ### gif Animated GIF demuxer. It accepts the following options: ### min_delay Set the minimum valid delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 2. ### max_gif_delay Set the maximum valid delay between frames in hundredth of seconds. Range is 0 to 65535. Default value is 65535 (nearly eleven minutes), the maximum value allowed by the specification. ### default_delay Set the default delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 10. ### ignore_loop GIF files can contain information to loop a certain number of times (or infinitely). If **ignore_loop **is set to 1, then the loop setting from the input will be ignored and looping will not occur. If set to 0, then looping will occur and will cycle the number of times according to the GIF. Default value is 1. For example, with the overlay filter, place an infinitely looping GIF over another video: ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv Note that in the above example the shortest option for overlay filter is used to end the output video at the length of the shortest input file, which in this case is _input.mp4_ as the GIF in this example loops infinitely. ### hls 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: ### live_start_index segment index to start live streams at (negative values are from the end). ### prefer_x_start prefer to use #EXT-X-START if it's in playlist instead of live_start_index. ### allowed_extensions ',' separated list of file extensions that hls is allowed to access. ### max_reload Maximum number of times a insufficient list is attempted to be reloaded. Default value is 1000. ### m3u8_hold_counters The maximum number of times to load m3u8 when it refreshes without new segments. Default value is 1000. ### http_persistent Use persistent HTTP connections. Applicable only for HTTP streams. Enabled by default. ### http_multiple Use multiple HTTP connections for downloading HTTP segments. Enabled by default for HTTP/1.1 servers. ### http_seekable Use HTTP partial requests for downloading HTTP segments. 0 = disable, 1 = enable, -1 = auto, Default is auto. ### seg_format_options Set options for the demuxer of media segments using a list of key=value pairs separated by ":". ### seg_max_retry Maximum number of times to reload a segment on error, useful when segment skip on network error is not desired. Default value is 0. ### image2 Image file demuxer. This demuxer reads from a list of image files specified by a pattern. The syntax and meaning of the pattern is specified by the option _pattern_type_. The pattern may contain a suffix which is used to automatically determine the format of the images contained in the files. The size, the pixel format, and the format of each image must be the same for all the files in the sequence. This demuxer accepts the following options: ### framerate Set the frame rate for the video stream. It defaults to 25. ### loop If set to 1, loop over the input. Default value is 0. ### pattern_type Select the pattern type used to interpret the provided filename. _pattern_type_ accepts one of the following values. **none** Disable pattern matching, therefore the video will only contain the specified image. You should use this option if you do not want to create sequences from multiple images and your filenames may contain special pattern characters. **sequence** Select a sequence pattern type, used to specify a sequence of files indexed by sequential numbers. A sequence pattern may contain the string "%d" or "%0_N_d", which specifies the position of the characters representing a sequential number in each filename matched by the pattern. If the form "%d0_N_d" 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 "%0_N_d", 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 "%0_N_d", for example to convert a single image file _img.jpeg_ you can employ the command: ffmpeg -i img.jpeg img.png **glob** Select a glob wildcard pattern type. The pattern is interpreted like a glob() pattern. This is only selectable if libavformat was compiled with globbing support. **glob_sequence _**(deprecated,_ _will_ _be_ _removed)_ Select a mixed glob wildcard/sequence pattern. If your version of libavformat was compiled with globbing support, and the provided pattern contains at least one glob meta character among "%*?[]{}" that is preceded by an unescaped "%", the pattern is interpreted like a glob() pattern, otherwise it is interpreted like a sequence pattern. All glob special characters "%*?[]{}" must be prefixed with "%". To escape a literal "%" you shall use "%%". For example the pattern "foo-%*.jpeg" will match all the filenames prefixed by "foo-" and terminating with ".jpeg", and "foo-%?%?%?.jpeg" will match all the filenames prefixed with "foo-", followed by a sequence of three characters, and terminating with ".jpeg". This pattern type is deprecated in favor of _glob_ and _sequence_. Default value is _glob_sequence_. ### pixel_format Set the pixel format of the images to read. If not specified the pixel format is guessed from the first image file in the sequence. ### start_number Set the index of the file matched by the image file pattern to start to read from. Default value is 0. ### start_number_range Set the index interval range to check when looking for the first image file in the sequence, starting from _start_number_. Default value is 5. ### ts_from_file If set to 1, will set frame timestamp to modification time of image file. Note that monotonity of timestamps is not provided: images go in the same order as without this option. Default value is 0. If set to 2, will set frame timestamp to the modification time of the image file in nanosecond precision. ### video_size Set the video size of the images to read. If not specified the video size is guessed from the first image file in the sequence. ### export_path_metadata 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: **lavf.image2dec.source_path** Corresponds to the full path to the input file being read. **lavf.image2dec.source_basename** 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 ### libgme The Game Music Emu library is a collection of video game music file emulators. See <****> for more information. It accepts the following options: ### track_index 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. ### sample_rate Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100. ### max_size _(bytes) 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. ### libmodplug ModPlug based module demuxer See <****> 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: ### noise_reduction Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0. ### reverb_depth Set amount of reverb. Range 0-100. Default is 0. ### reverb_delay Set delay in ms, clamped to 40-250 ms. Default is 0. ### bass_amount Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0. ### bass_range Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0. ### surround_depth Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0. ### surround_delay Set surround delay in ms, clamped to 5-40 ms. Default is 0. ### max_size 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. ### video_stream_expr 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". ### video_stream Generate video stream. Can be 1 (on) or 0 (off). Default is 0. ### video_stream_w Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30. ### video_stream_h Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30. ### video_stream_ptxt 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 libopenmpt based module demuxer See <****> for more information. Some files have multiple subsongs (tracks) this can be set with the **subsong **option. It accepts the following options: ### subsong Set the subsong index. This can be either 'all', 'auto', or the index of the subsong. Subsong indexes start at 0. The default is 'auto'. The default value is to let libopenmpt choose. ### layout Set the channel layout. Valid values are 1, 2, and 4 channel layouts. The default value is STEREO. ### sample_rate Set the sample rate for libopenmpt to output. Range is from 1000 to INT_MAX. The value default is 48000. ### mov/mp4/3gp 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_drefs Enable loading of external tracks, disabled by default. Enabling this can theoretically leak information in some use cases. ### use_absolute_path Allows loading of external tracks via absolute paths, disabled by default. Enabling this poses a security risk. It should only be enabled if the source is known to be non- malicious. ### seek_streams_individually 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_editlist 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. ### advanced_editlist 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. ### ignore_chapters Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are only parsed when input is seekable. Default is false. ### use_mfra_for For seekable fragmented input, set fragment's starting timestamp from media fragment random access box, if present. Following options are available: **auto** Auto-detect whether to set mfra timestamps as PTS or DTS _(default)_ **dts **Set mfra timestamps as DTS **pts **Set mfra timestamps as PTS **0 **Don't use mfra box to set timestamps ### use_tfdt 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_all 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_xmp 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. ### activation_bytes 4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below. ### audible_fixed_key Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to specify. ### decryption_key 16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7). ### max_stts_delta 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 upto a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of "uint32" range. ### interleaved_read 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 ### mpegts MPEG-2 transport stream demuxer. This demuxer accepts the following options: ### resync_size Set size limit for looking up a new synchronization. Default value is 65536. ### skip_unknown_pmt Skip PMTs for programs not defined in the PAT. Default value is 0. ### fix_teletext_pts Override teletext packet PTS and DTS values with the timestamps calculated from the PCR of the first program which the teletext stream is part of and is not discarded. Default value is 1, set this option to 0 if you want your teletext packet PTS and DTS values untouched. ### ts_packetsize Output option carrying the raw packet size in bytes. Show the detected raw packet size, cannot be set by the user. ### scan_all_pmts Scan and combine all PMTs. The value is an integer with value from -1 to 1 (-1 means automatic setting, 1 means enabled, 0 means disabled). Default value is -1. ### merge_pmt_versions Re-use existing streams when a PMT's version is updated and elementary streams move to different PIDs. Default value is 0. ### max_packet_size 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. ### mpjpeg MJPEG encapsulated in multi-part MIME demuxer. This demuxer allows reading of MJPEG, where each frame is represented as a part of multipart/x-mixed-replace stream. ### strict_mime_boundary Default implementation applies a relaxed standard to multi-part MIME boundary detection, to prevent regression with numerous existing endpoints not generating a proper MIME MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check of the boundary value. ### rawvideo Raw video demuxer. This demuxer allows one to read raw video data. Since there is no header specifying the assumed video parameters, the user must specify them in order to be able to decode the data correctly. This demuxer accepts the following options: ### framerate Set input video frame rate. Default value is 25. ### pixel_format Set the input video pixel format. Default value is "yuv420p". ### video_size Set the input video size. This value must be specified explicitly. For example to read a rawvideo file _input.raw_ with **ffplay**, assuming a pixel format of "rgb24", a video size of "320x240", and a frame rate of 10 images per second, use the command: ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw ### sbg SBaGen script demuxer. This demuxer reads the script language used by SBaGen <****> to generate binaural beats sessions. A SBG script looks like that: -SE a: 300-2.5/3 440+4.5/0 b: 300-2.5/0 440+4.5/3 off: - NOW == a +0:07:00 == b +0:14:00 == a +0:21:00 == b +0:30:00 off A SBG script can mix absolute and relative timestamps. If the script uses either only absolute timestamps (including the script start time) or only relative ones, then its layout is fixed, and the conversion is straightforward. On the other hand, if the script mixes both kind of timestamps, then the _NOW_ reference for relative timestamps will be taken from the current time of day at the time the script is read, and the script layout will be frozen according to that reference. That means that if the script is directly played, the actual times will match the absolute timestamps up to the sound controller's clock accuracy, but if the user somehow pauses the playback or seeks, all times will be shifted accordingly. ### tedcaptions JSON captions used for <****>. TED does not provide links to the captions, but they can be guessed from the page. The file _tools/bookmarklets.html_ from the FFmpeg source tree contains a bookmarklet to expose them. This demuxer accepts the following option: ### start_time Set the start time of the TED talk, in milliseconds. The default is 15000 (15s). It is used to sync the captions with the downloadable videos, because they include a 15s intro. Example: convert the captions to a format most players understand: ffmpeg -i talk1-en.srt ### vapoursynth 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: ### max_script_size The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of scripts that can be read. Default is 1 MiB. ## MUXERS 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. ### a64 A64 muxer for Commodore 64 video. Accepts a single "a64_multi" or "a64_multi5" codec video stream. ### adts Audio Data Transport Stream muxer. It accepts a single AAC stream. _Options_ It accepts the following options: **write_id3v2 _**bool_ Enable to write ID3v2.4 tags at the start of the stream. Default is disabled. **write_apetag _**bool_ Enable to write APE tags at the end of the stream. Default is disabled. **write_mpeg2 _**bool_ Enable to set MPEG version bit in the ADTS frame header to 1 which indicates MPEG-2. Default is 0, which indicates MPEG-4. ### aiff Audio Interchange File Format muxer. _Options_ It accepts the following options: ### write_id3v2 Enable ID3v2 tags writing when set to 1. Default is 0 (disabled). ### id3v2_version Select ID3v2 version to write. Currently only version 3 and 4 (aka. ID3v2.3 and ID3v2.4) are supported. The default is version 4. ### alp Muxer for audio of High Voltage Software's Lego Racers game. It accepts a single ADPCM_IMA_ALP stream with no more than 2 channels nor a sample rate greater than 44100 Hz. Extensions: tun, pcm _Options_ It accepts the following options: **type _**type_ Set file type. **tun **Set file type as music. Must have a sample rate of 22050 Hz. **pcm **Set file type as sfx. **auto** Set file type as per output file extension. ".pcm" results in type "pcm" else type "tun" is set. _(default)_ ### asf Advanced Systems Format muxer. Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this muxer too. _Options_ It accepts the following options: ### packet_size Set the muxer packet size. By tuning this setting you may reduce data fragmentation or muxer overhead depending on your source. Default value is 3200, minimum is 100, maximum is 64k. ### avi Audio Video Interleaved muxer. _Options_ It accepts the following options: ### reserve_index_space Reserve the specified amount of bytes for the OpenDML master index of each stream within the file header. By default additional master indexes are embedded within the data packets if there is no space left in the first master index and are linked together as a chain of indexes. This index structure can cause problems for some use cases, e.g. third- party software strictly relying on the OpenDML index specification or when file seeking is slow. Reserving enough index space in the file header avoids these problems. The required index space depends on the output file size and should be about 16 bytes per gigabyte. When this option is omitted or set to zero the necessary index space is guessed. ### write_channel_mask 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**). ### flipped_raw_rgb If set to true, store positive height for raw RGB bitmaps, which indicates bitmap is stored bottom-up. Note that this option does not flip the bitmap which has to be done manually beforehand, e.g. by using the vflip filter. Default is _false_ and indicates bitmap is stored top down. ### chromaprint Chromaprint fingerprinter. This muxer feeds audio data to the Chromaprint library, which generates a fingerprint for the provided audio data. See <****> It takes a single signed native-endian 16-bit raw audio stream of at most 2 channels. _Options_ ### silence_threshold 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. ### algorithm Version of algorithm to fingerprint with. Range is 0 to 4. Version 3 enables silence detection. Default is 1. ### fp_format Format to output the fingerprint as. Accepts the following options: **raw **Binary raw fingerprint **compressed** Binary compressed fingerprint **base64** Base64 compressed fingerprint _(default)_ ### crc CRC (Cyclic Redundancy Check) testing format. This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC. The output of the muxer consists of a single line of the form: CRC=0x_CRC_, where _CRC_ is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames. See also the **framecrc **muxer. _Examples_ For example to compute the CRC of the input, and store it in the file _out.crc_: ffmpeg -i INPUT -f crc out.crc You can print the CRC to stdout with the command: ffmpeg -i INPUT -f crc - You can select the output format of each frame with **ffmpeg **by specifying the audio and video codec and format. For example to compute the CRC of the input audio converted to PCM unsigned 8-bit and the input video converted to MPEG-2 video, use the command: ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - ### dash Dynamic Adaptive Streaming over HTTP (DASH) muxer that creates segments and manifest files according to the MPEG-DASH standard ISO/IEC 23009-1:2014. For more information see: • ISO DASH Specification: <****> • WebM DASH Specification: <****> It creates a MPD manifest file and segment files for each stream. The segment filename might contain pre-defined identifiers used with SegmentTemplate as defined in section 5.3.9.4.4 of the standard. Available identifiers are "$RepresentationID$", "$Number$", "$Bandwidth$" and "$Time$". In addition to the standard identifiers, an ffmpeg- specific "$ext$" identifier is also supported. When specified ffmpeg will replace $ext$ in the file name with muxing format's extensions such as mp4, webm etc., ffmpeg -re -i -map 0 -map 0 -c:a libfdk_aac -c:v libx264 \ -b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline \ -profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0 \ -b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1 \ -window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a" \ -f dash /path/to/out.mpd **seg_duration _**duration_ Set the segment length in seconds (fractional value can be set). The value is treated as average segment duration when _use_template_ is enabled and _use_timeline_ is disabled and as minimum segment duration for all the other use cases. **frag_duration _**duration_ Set the length in seconds of fragments within segments (fractional value can be set). **frag_type _**type_ Set the type of interval for fragmentation. **window_size _**size_ Set the maximum number of segments kept in the manifest. **extra_window_size _**size_ Set the maximum number of segments kept outside of the manifest before removing from disk. **remove_at_exit _**remove_ Enable (1) or disable (0) removal of all segments when finished. **use_template _**template_ Enable (1) or disable (0) use of SegmentTemplate instead of SegmentList. **use_timeline _**timeline_ Enable (1) or disable (0) use of SegmentTimeline in SegmentTemplate. **single_file _**single_file_ Enable (1) or disable (0) storing all segments in one file, accessed using byte ranges. **single_file_name _**file_name_ DASH-templated name to be used for baseURL. Implies _single_file_ set to "1". In the template, "$ext$" is replaced with the file name extension specific for the segment format. **init_seg_name _**init_name_ DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format. **media_seg_name _**segment_name_ DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.$ext$". "$ext$" is replaced with the file name extension specific for the segment format. **utc_timing_url _**utc_url_ URL of the page that will return the UTC timestamp in ISO format. Example: "" **method _**method_ Use the given HTTP method to create output files. Generally set to PUT or POST. **http_user_agent _**user_agent_ Override User-Agent field in HTTP header. Applicable only for HTTP output. **http_persistent _**http_persistent_ Use persistent HTTP connections. Applicable only for HTTP output. **hls_playlist _**hls_playlist_ Generate HLS playlist files as well. The master playlist is generated with the filename _hls_master_name_. One media playlist file is generated for each stream with filenames media_0.m3u8, media_1.m3u8, etc. **hls_master_name _**file_name_ HLS master playlist name. Default is "master.m3u8". **streaming _**streaming_ Enable (1) or disable (0) chunk streaming mode of output. In chunk streaming mode, each frame will be a moof fragment which forms a chunk. **adaptation_sets _**adaptation_sets_ Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c id=y,streams=d,e" with x and y being the IDs of the adaptation sets and a,b,c,d and e are the indices of the mapped streams. To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be used as stream identifier instead of IDs. When no assignment is defined, this defaults to an AdaptationSet for each stream. Optional syntax is "id=x,seg_duration=x,frag_duration=x,frag_type=type,descriptor=descriptor_string,streams=a,b,c id=y,seg_duration=y,frag_type=type,streams=d,e" and so on, descriptor is useful to the scheme defined by ISO/IEC 23009-1:2014/Amd.2:2015. For example, -adaptation_sets "id=0,descriptor=,streams=v". Please note that descriptor string should be a self-closing xml tag. seg_duration, frag_duration and frag_type override the global option values for each adaptation set. For example, -adaptation_sets "id=0,seg_duration=2,frag_duration=1,frag_type=duration,streams=v id=1,seg_duration=2,frag_type=none,streams=a" type_id marks an adaptation set as containing streams meant to be used for Trick Mode for the referenced adaptation set. For example, -adaptation_sets "id=0,seg_duration=2,frag_type=none,streams=0 id=1,seg_duration=10,frag_type=none,trick_id=0,streams=1" **timeout _**timeout_ Set timeout for socket I/O operations. Applicable only for HTTP output. **index_correction _**index_correction_ Enable (1) or Disable (0) segment index correction logic. Applicable only when _use_template_ is enabled and _use_timeline_ is disabled. When enabled, the logic monitors the flow of segment indexes. If a streams's segment index value is not at the expected real time position, then the logic corrects that index value. Typically this logic is needed in live streaming use cases. The network bandwidth fluctuations are common during long run streaming. Each fluctuation can cause the segment indexes fall behind the expected real time position. **format_options _**options_list_ Set container format (mp4/webm) options using a ":" separated list of key=value parameters. Values containing ":" special characters must be escaped. **global_sidx _**global_sidx_ Write global SIDX atom. Applicable only for single file, mp4 output, non-streaming mode. **dash_segment_type _**dash_segment_type_ Possible values: **auto** If this flag is set, the dash segment files format will be selected based on the stream codec. This is the default mode. **mp4 **If this flag is set, the dash segment files will be in in ISOBMFF format. **webm** If this flag is set, the dash segment files will be in in WebM format. **ignore_io_errors _**ignore_io_errors_ Ignore IO errors during open and write. Useful for long-duration runs with network output. **lhls _**lhls_ Enable Low-latency HLS(LHLS). Adds #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 in 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 <****> **ldash _**ldash_ Enable Low-latency Dash by constraining the presence and values of some elements. **master_m3u8_publish_rate _**master_m3u8_publish_rate_ Publish master playlist repeatedly every after specified number of segment intervals. **write_prft _**write_prft_ Write Producer Reference Time elements on supported streams. This also enables writing prft boxes in the underlying muxer. Applicable only when the _utc_url_ option is enabled. It's set to auto by default, in which case the muxer will attempt to enable it only in modes that require it. **mpd_profile _**mpd_profile_ Set one or more manifest profiles. **http_opts _**http_opts_ A :-separated list of key=value options to pass to the underlying HTTP protocol. Applicable only for HTTP output. **target_latency _**target_latency_ Set an intended target latency in seconds (fractional value can be set) for serving. Applicable only when _streaming_ and _write_prft_ options are enabled. This is an informative fields clients can use to measure the latency of the service. **min_playback_rate _**min_playback_rate_ Set the minimum playback rate indicated as appropriate for the purposes of automatically adjusting playback latency and buffer occupancy during normal playback by clients. **max_playback_rate _**max_playback_rate_ Set the maximum playback rate indicated as appropriate for the purposes of automatically adjusting playback latency and buffer occupancy during normal playback by clients. **update_period _**update_period_ Set the mpd update period ,for dynamic content. The unit is second. ### fifo The fifo pseudo-muxer allows the separation of encoding and muxing by using first-in-first- out queue and running the actual muxer in a separate thread. This is especially useful in combination with the **tee **muxer and can be used to send data to several destinations with different reliability/writing speed/latency. API users should be aware that callback functions (interrupt_callback, io_open and io_close) used within its AVFormatContext must be thread-safe. The behavior of the fifo muxer if the queue fills up or if the output fails is selectable, • output can be transparently restarted with configurable delay between retries based on real time or time of the processed stream. • encoding can be blocked during temporary failure, or continue transparently dropping packets in case fifo queue fills up. ### fifo_format Specify the format name. Useful if it cannot be guessed from the output name suffix. ### queue_size Specify size of the queue (number of packets). Default value is 60. ### format_opts Specify format options for the underlying muxer. Muxer options can be specified as a list of _key_=_value_ pairs separated by ':'. **drop_pkts_on_overflow _**bool_ If set to 1 (true), in case the fifo queue fills up, packets will be dropped rather than blocking the encoder. This makes it possible to continue streaming without delaying the input, at the cost of omitting part of the stream. By default this option is set to 0 (false), so in such cases the encoder will be blocked until the muxer processes some of the packets and none of them is lost. **attempt_recovery _**bool_ If failure occurs, attempt to recover the output. This is especially useful when used with network output, since it makes it possible to restart streaming transparently. By default this option is set to 0 (false). ### max_recovery_attempts Sets maximum number of successive unsuccessful recovery attempts after which the output fails permanently. By default this option is set to 0 (unlimited). **recovery_wait_time _**duration_ Waiting time before the next recovery attempt after previous unsuccessful recovery attempt. Default value is 5 seconds. **recovery_wait_streamtime _**bool_ If set to 0 (false), the real time is used when waiting for the recovery attempt (i.e. the recovery will be attempted after at least recovery_wait_time seconds). If set to 1 (true), the time of the processed stream is taken into account instead (i.e. the recovery will be attempted after at least _recovery_wait_time_ seconds of the stream is omitted). By default, this option is set to 0 (false). **recover_any_error _**bool_ If set to 1 (true), recovery will be attempted regardless of type of the error causing the failure. By default this option is set to 0 (false) and in case of certain (usually permanent) errors the recovery is not attempted even when _attempt_recovery_ is set to 1. **restart_with_keyframe _**bool_ Specify whether to wait for the keyframe after recovering from queue overflow or failure. This option is set to 0 (false) by default. **timeshift _**duration_ Buffer the specified amount of packets and delay writing the output. Note that _queue_size_ must be big enough to store the packets for timeshift. At the end of the input the fifo buffer is flushed at realtime speed. _Examples_ • Stream something to rtmp server, continue processing the stream at real-time rate even in case of temporary failure (network outage) and attempt to recover streaming every second indefinitely. ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name ### flv Adobe Flash Video Format muxer. This muxer accepts the following options: **flvflags _**flags_ Possible values: **aac_seq_header_detect** Place AAC sequence header based on audio stream data. **no_sequence_end** Disable sequence end tag. **no_metadata** Disable metadata tag. **no_duration_filesize** Disable duration and filesize in metadata when they are equal to zero at the end of stream. (Be used to non-seekable living stream). **add_keyframe_index** Used to facilitate seeking; particularly for HTTP pseudo streaming. ### framecrc Per-packet CRC (Cyclic Redundancy Check) testing format. This muxer computes and prints the Adler-32 CRC for each audio and video packet. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC. The output of the muxer consists of a line for each audio and video packet of the form: , , , , , 0x _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. ### framehash Per-packet hash testing format. This muxer computes and prints a cryptographic hash for each audio and video packet. This can be used for packet-by-packet equality checks without having to individually do a binary comparison on each. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash, but the output of explicit conversions to other codecs can also be used. It uses the SHA-256 cryptographic hash function by default, but supports several other algorithms. The output of the muxer consists of a line for each audio and video packet of the form: , , , , , _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. ### framemd5 Per-packet MD5 testing format. This is a variant of the **framehash **muxer. Unlike that muxer, it defaults to using the MD5 hash function. _Examples_ To compute the MD5 hash of the audio and video frames in _INPUT_, converted to raw audio and video packets, and store it in the file _out.md5_: ffmpeg -i INPUT -f framemd5 out.md5 To print the information to stdout, use the command: ffmpeg -i INPUT -f framemd5 - See also the **framehash **and **md5 **muxers. ### gif Animated GIF muxer. It accepts the following options: ### loop Set the number of times to loop the output. Use -1 for no loop, 0 for looping indefinitely (default). ### final_delay Force the delay (expressed in centiseconds) after the last frame. Each frame ends with a delay until the next frame. The default is -1, which is a special value to tell the muxer to re-use the previous delay. In case of a loop, you might want to customize this value to mark a pause for instance. For example, to encode a gif looping 10 times, with a 5 seconds delay between the loops: ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif Note 1: if you wish to extract the frames into separate GIF files, you need to force the **image2 **muxer: ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif" Note 2: the GIF format has a very large time base: the delay between two frames can therefore not be smaller than one centi second. ### hash Hash testing format. This muxer computes and prints a cryptographic hash of all the input audio and video frames. This can be used for equality checks without having to do a complete binary comparison. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash, but the output of explicit conversions to other codecs can also be used. Timestamps are ignored. It uses the SHA-256 cryptographic hash function by default, but supports several other algorithms. The output of the muxer consists of a single line of the form: _algo_=_hash_, where _algo_ is a short string representing the hash function used, and _hash_ is a hexadecimal number representing the computed hash. **hash _**algorithm_ Use the cryptographic hash function specified by the string _algorithm_. Supported values include "MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256" (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32" and "adler32". _Examples_ To compute the SHA-256 hash of the input converted to raw audio and video, and store it in the file _out.sha256_: ffmpeg -i INPUT -f hash out.sha256 To print an MD5 hash to stdout use the command: ffmpeg -i INPUT -f hash -hash md5 - See also the **framehash **muxer. ### hls Apple HTTP Live Streaming muxer that segments MPEG-TS according to the HTTP Live Streaming (HLS) specification. It creates a playlist file, and one or more segment files. The output filename specifies the playlist filename. By default, the muxer creates a file for each segment produced. These files have the same name as the playlist, followed by a sequential number and a .ts extension. Make sure to require a closed GOP when encoding and to set the GOP size to fit your segment time constraint. For example, to convert an input file with **ffmpeg**: ffmpeg -i in.mkv -c:v h264 -flags +cgop -g 30 -hls_time 1 out.m3u8 This example will produce the playlist, _out.m3u8_, and segment files: _out0.ts_, _out1.ts_, _out2.ts_, etc. See also the **segment **muxer, which provides a more generic and flexible implementation of a segmenter, and can be used to perform HLS segmentation. _Options_ This muxer supports the following options: **hls_init_time _**duration_ 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)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown) 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" **hls_time _**duration_ 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)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown) manual**. Segment will be cut on the next key frame after this time has passed. **hls_list_size _**size_ Set the maximum number of playlist entries. If set to 0 the list file will contain all the segments. Default value is 5. **hls_delete_threshold _**size_ 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. ### hls_start_number_source 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: **generic (default)** Set the starting sequence numbers according to _start_number_ option value. **epoch** The start number will be the seconds since epoch (1970-01-01 00:00:00) **epoch_us** The start number will be the microseconds since epoch (1970-01-01 00:00:00) **datetime** The start number will be based on the current date/time as YYYYmmddHHMMSS. e.g. 20161231235759. **start_number _**number_ 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. **hls_allow_cache _**allowcache_ Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments. **hls_base_url _**baseurl_ Append _baseurl_ to every entry in the playlist. Useful to generate playlists with absolute paths. Note that the playlist sequence number must be unique for each segment and it is not to be confused with the segment filename sequence number which can be cyclic, for example if the **wrap **option is specified. **hls_segment_filename _**filename_ Set the segment filename. Unless "hls_flags single_file" is set, _filename_ is used as a string format with the segment number: ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8 This example will produce the playlist, _out.m3u8_, and segment files: _file000.ts_, _file001.ts_, _file002.ts_, etc. _filename_ may contain full path or relative path specification, but only the file name part without any path info will be contained in the m3u8 segment list. Should a relative path be specified, the path of the created segment files will be relative to the current working directory. When strftime_mkdir is set, the whole expanded value of _filename_ will be written into the m3u8 segment list. When "var_stream_map" is set with two or more variant streams, the _filename_ pattern must contain the string "%v", this string specifies the position of variant stream index in the generated segment file names. ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \ -hls_segment_filename 'file_%v_%03d.ts' out_%v.m3u8 This example will produce the playlists segment file sets: _file_0_000.ts_, _file_0_001.ts_, _file_0_002.ts_, etc. and _file_1_000.ts_, _file_1_001.ts_, _file_1_002.ts_, etc. The string "%v" may be present in the filename or in the last directory name containing the file, but only in one of them. (Additionally, %v may appear multiple times in the last sub-directory or filename.) If the string %v is present in the directory name, then sub-directories are created after expanding the directory name pattern. This enables creation of segments corresponding to different variant streams in subdirectories. ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \ -hls_segment_filename 'vs%v/file_%03d.ts' vs%v/out.m3u8 This example will produce the playlists segment file sets: _vs0/file_000.ts_, _vs0/file_001.ts_, _vs0/file_002.ts_, etc. and _vs1/file_000.ts_, _vs1/file_001.ts_, _vs1/file_002.ts_, etc. ### strftime Use **strftime() **on _filename_ to expand the segment filename with localtime. The segment number is also available in this mode, but to use it, you need to specify second_level_segment_index hls_flag and %%d will be the specifier. ffmpeg -i in.nut -strftime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8 This example will produce the playlist, _out.m3u8_, and segment files: _file-20160215-1455569023.ts_, _file-20160215-1455569024.ts_, etc. Note: On some systems/environments, the %s specifier is not available. See strftime() documentation. ffmpeg -i in.nut -strftime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8 This example will produce the playlist, _out.m3u8_, and segment files: _file-20160215-0001.ts_, _file-20160215-0002.ts_, etc. ### strftime_mkdir Used together with -strftime_mkdir, it will create all subdirectories which is expanded in _filename_. ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8 This example will create a directory 201560215 (if it does not exist), and then produce the playlist, _out.m3u8_, and segment files: _20160215/file-20160215-1455569023.ts_, _20160215/file-20160215-1455569024.ts_, etc. ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8 This example will create a directory hierarchy 2016/02/15 (if any of them do not exist), and then produce the playlist, _out.m3u8_, and segment files: _2016/02/15/file-20160215-1455569023.ts_, _2016/02/15/file-20160215-1455569024.ts_, etc. **hls_segment_options _**options_list_ Set output format options using a :-separated list of key=value parameters. Values containing ":" special characters must be escaped. **hls_key_info_file _**key_info_file_ Use the information in _key_info_file_ for segment encryption. The first line of _key_info_file_ specifies the key URI written to the playlist. The key URL is used to access the encryption key during playback. The second line specifies the path to the key file used to obtain the key during the encryption process. The key file is read as a single packed array of 16 octets in binary format. The optional third line specifies the initialization vector (IV) as a hexadecimal string to be used instead of the segment sequence number (default) for encryption. Changes to _key_info_file_ will result in segment encryption with the new key/IV and an entry in the playlist for the new key URI/IV if "hls_flags periodic_rekey" is enabled. Key info file format: (optional) Example key URIs: /path/to/file.key file.key Example key file paths: file.key /path/to/file.key Example IV: 0123456789ABCDEF0123456789ABCDEF Key info file example: /path/to/file.key 0123456789ABCDEF0123456789ABCDEF Example shell script: #!/bin/sh BASE_URL=${1:-'.'} openssl rand 16 > file.key echo $BASE_URL/file.key > file.keyinfo echo file.key >> file.keyinfo echo $(openssl rand -hex 16) >> file.keyinfo ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \ -hls_key_info_file file.keyinfo out.m3u8 ### -hls_enc _ 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. ### -hls_enc_key _ 16-octet key to encrypt the segments, by default it is randomly generated. ### -hls_enc_key_url _ If set, _keyurl_ is prepended instead of _baseurl_ to the key filename in the playlist. ### -hls_enc_iv _ 16-octet initialization vector for every segment instead of the autogenerated ones. **hls_segment_type _**flags_ Possible values: **mpegts** Output segment files in MPEG-2 Transport Stream format. This is compatible with all HLS versions. **fmp4** Output segment files in fragmented MP4 format, similar to MPEG-DASH. fmp4 files may be used in HLS version 7 and above. **hls_fmp4_init_filename _**filename_ Set filename to the fragment files header file, default filename is _init.mp4_. Use "-strftime 1" on _filename_ to expand the segment filename with localtime. ffmpeg -i in.nut -hls_segment_type fmp4 -strftime 1 -hls_fmp4_init_filename "%s_init.mp4" out.m3u8 This will produce init like this _1602678741_init.mp4_ ### hls_fmp4_init_resend 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. **hls_flags _**flags_ Possible values: **single_file** If this flag is set, the muxer will store all segments in a single MPEG-TS file, and will use byte ranges in the playlist. HLS playlists generated with this way will have the version number 4. For example: ffmpeg -i in.nut -hls_flags single_file out.m3u8 Will produce the playlist, _out.m3u8_, and a single segment file, _out.ts_. **delete_segments** Segment files removed from the playlist are deleted after a period of time equal to the duration of the segment plus the duration of the playlist. **append_list** Append new segments into the end of old segment list, and remove the "#EXT-X-ENDLIST" from the old segment list. **round_durations** 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. **discont_start** Add the "#EXT-X-DISCONTINUITY" tag to the playlist, before the first segment's information. **omit_endlist** Do not append the "EXT-X-ENDLIST" tag at the end of the playlist. **periodic_rekey** 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. **independent_segments** Add the "#EXT-X-INDEPENDENT-SEGMENTS" to playlists that has video segments and when all the segments of that playlist are guaranteed to start with a Key frame. **iframes_only** Add the "#EXT-X-I-FRAMES-ONLY" to playlists that has video segments and can play only I-frames in the "#EXT-X-BYTERANGE" mode. **split_by_time** Allow segments to start on frames other than keyframes. This improves behavior on some players when the time between keyframes is inconsistent, but may make things worse on others, and can cause some oddities during seeking. This flag should be used with the "hls_time" option. **program_date_time** Generate "EXT-X-PROGRAM-DATE-TIME" tags. **second_level_segment_index** Makes it possible to use segment indexes as %%d in hls_segment_filename expression besides date/time values when strftime is on. To get fixed width numbers with trailing zeroes, %%0xd format is available where x is the required width. **second_level_segment_size** Makes it possible to use segment sizes (counted in bytes) as %%s in hls_segment_filename expression besides date/time values when strftime is on. To get fixed width numbers with trailing zeroes, %%0xs format is available where x is the required width. **second_level_segment_duration** Makes it possible to use segment duration (calculated in microseconds) as %%t in hls_segment_filename expression besides date/time values when strftime is on. To get fixed width numbers with trailing zeroes, %%0xt format is available where x is the required width. ffmpeg -i sample.mpeg \ -f hls -hls_time 3 -hls_list_size 5 \ -hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \ -strftime 1 -strftime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8 This will produce segments like this: _segment_20170102194334_0003_00122200_0000003000000.ts_, _segment_20170102194334_0004_00120072_0000003000000.ts_ etc. **temp_file** Write segment data to filename.tmp and rename to filename only once the segment is complete. A webserver serving up segments can be configured to reject requests to *.tmp to prevent access to in-progress segments before they have been added to the m3u8 playlist. This flag also affects how m3u8 playlist files are created. If this flag is set, all playlist files will written into temporary file and renamed after they are complete, similarly as segments are handled. But playlists with "file" protocol and with type ("hls_playlist_type") other than "vod" are always written into temporary file regardless of this flag. Master playlist files ("master_pl_name"), if any, with "file" protocol, are always written into temporary file regardless of this flag if "master_pl_publish_rate" value is other than zero. ### hls_playlist_type event Emit "#EXT-X-PLAYLIST-TYPE:EVENT" in the m3u8 header. Forces **hls_list_size **to 0; the playlist can only be appended to. ### hls_playlist_type vod Emit "#EXT-X-PLAYLIST-TYPE:VOD" in the m3u8 header. Forces **hls_list_size **to 0; the playlist must not change. ### method Use the given HTTP method to create the hls files. ffmpeg -re -i in.ts -f hls -method PUT This example will upload all the mpegts segment files to the HTTP server using the HTTP PUT method, and update the m3u8 files every "refresh" times using the same method. Note that the HTTP server must support the given method for uploading files. ### http_user_agent Override User-Agent field in HTTP header. Applicable only for HTTP output. ### var_stream_map Map string which specifies how to group the audio, video and subtitle streams into different variant streams. The variant stream groups are separated by space. Expected string format is like this "a:0,v:0 a:1,v:1 ....". Here a:, v:, s: are the keys to specify audio, video and subtitle streams respectively. Allowed values are 0 to 9 (limited just based on practical usage). When there are two or more variant streams, the output filename pattern must contain the string "%v", this string specifies the position of variant stream index in the output media playlist filenames. The string "%v" may be present in the filename or in the last directory name containing the file. If the string is present in the directory name, then sub-directories are created after expanding the directory name pattern. This enables creation of variant streams in subdirectories. ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \ This example creates two hls variant streams. The first variant stream will contain video stream of bitrate 1000k and audio stream of bitrate 64k and the second variant stream will contain video stream of bitrate 256k and audio stream of bitrate 32k. Here, two media playlist with file names out_0.m3u8 and out_1.m3u8 will be created. If you want something meaningful text instead of indexes in result names, you may specify names for each or some of the variants as in the following example. ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0,name:my_hd v:1,a:1,name:my_sd" \ This example creates two hls variant streams as in the previous one. But here, the two media playlist with file names out_my_hd.m3u8 and out_my_sd.m3u8 will be created. ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k \ -map 0:v -map 0:a -map 0:v -f hls -var_stream_map "v:0 a:0 v:1" \ This example creates three hls variant streams. The first variant stream will be a video only stream with video bitrate 1000k, the second variant stream will be an audio only stream with bitrate 64k and the third variant stream will be a video only stream with bitrate 256k. Here, three media playlist with file names out_0.m3u8, out_1.m3u8 and out_2.m3u8 will be created. ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \ This example creates the variant streams in subdirectories. Here, the first media playlist is created at _ and the second one at _. 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 \ This example creates two audio only and two video only variant streams. In addition to the #EXT-X-STREAM-INF tag for each variant stream in the master playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams and they are mapped to the two video only variant streams with audio group names 'aud_low' and 'aud_high'. By default, a single hls variant containing all the encoded streams is created. ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \ -map 0:a -map 0:a -map 0:v -f hls \ -var_stream_map "a:0,agroup:aud_low,default:yes a:1,agroup:aud_low v:0,agroup:aud_low" \ -master_pl_name master.m3u8 \ This example creates two audio only and one video only variant streams. In addition to the #EXT-X-STREAM-INF tag for each variant stream in the master playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams and they are mapped to the one video only variant streams with audio group name 'aud_low', and the audio group have default stat is NO or YES. By default, a single hls variant containing all the encoded streams is created. ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \ -map 0:a -map 0:a -map 0:v -f hls \ -var_stream_map "a:0,agroup:aud_low,default:yes,language:ENG a:1,agroup:aud_low,language:CHN v:0,agroup:aud_low" \ -master_pl_name master.m3u8 \ This example creates two audio only and one video only variant streams. In addition to the #EXT-X-STREAM-INF tag for each variant stream in the master playlist, #EXT-X-MEDIA tag is also added for the two audio only variant streams and they are mapped to the one video only variant streams with audio group name 'aud_low', and the audio group have default stat is NO or YES, and one audio have and language is named ENG, the other audio language is named CHN. By default, a single hls variant containing all the encoded streams is created. ffmpeg -y -i input_with_subtitle.mkv \ -b:v:0 5250k -c:v h264 -pix_fmt yuv420p -profile:v main -level 4.1 \ -b:a:0 256k \ -c:s webvtt -c:a mp2 -ar 48000 -ac 2 -map 0:v -map 0:a:0 -map 0:s:0 \ -f hls -var_stream_map "v:0,a:0,s:0,sgroup:subtitle" \ -master_pl_name master.m3u8 -t 300 -hls_time 10 -hls_init_time 4 -hls_list_size \ 10 -master_pl_publish_rate 10 -hls_flags \ delete_segments+discont_start+split_by_time ./tmp/video.m3u8 This example adds "#EXT-X-MEDIA" tag with "TYPE=SUBTITLES" in the master playlist with webvtt subtitle group name 'subtitle'. Please make sure the input file has one text subtitle stream at least. ### cc_stream_map 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:,instreamid:,language: ....". 'ccgroup' and 'instreamid' are mandatory attributes. 'language' is an optional attribute. The closed captions groups configured using this option are mapped to different variant streams by providing the same 'ccgroup' name in the "var_stream_map" string. If "var_stream_map" is not set, then the first available ccgroup in "cc_stream_map" is mapped to the output variant stream. The examples for these two use cases are given below. ffmpeg -re -i in.ts -b:v 1000k -b:a 64k -a53cc 1 -f hls \ -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en" \ -master_pl_name master.m3u8 \ This example adds "#EXT-X-MEDIA" tag with "TYPE=CLOSED-CAPTIONS" in the master playlist with group name 'cc', language 'en' (english) and INSTREAM-ID 'CC1'. Also, it adds "CLOSED-CAPTIONS" attribute with group name 'cc' for the output variant stream. ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \ -a53cc:0 1 -a53cc:1 1\ -map 0:v -map 0:a -map 0:v -map 0:a -f hls \ -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en ccgroup:cc,instreamid:CC2,language:sp" \ -var_stream_map "v:0,a:0,ccgroup:cc v:1,a:1,ccgroup:cc" \ -master_pl_name master.m3u8 \ This example adds two "#EXT-X-MEDIA" tags with "TYPE=CLOSED-CAPTIONS" in the master playlist for the INSTREAM-IDs 'CC1' and 'CC2'. Also, it adds "CLOSED-CAPTIONS" attribute with group name 'cc' for the two output variant streams. ### master_pl_name Create HLS master playlist with the given name. ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 This example creates HLS master playlist with name master.m3u8 and it is published at ### master_pl_publish_rate Publish master play list repeatedly every after specified number of segment intervals. ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 \ -hls_time 2 -master_pl_publish_rate 30 This example creates HLS master playlist with name master.m3u8 and keep publishing it repeatedly every after 30 segments i.e. every after 60s. ### http_persistent Use persistent HTTP connections. Applicable only for HTTP output. ### timeout Set timeout for socket I/O operations. Applicable only for HTTP output. ### ignore_io_errors Ignore IO errors during open, write and delete. Useful for long-duration runs with network output. ### headers Set custom HTTP headers, can override built in default headers. Applicable only for HTTP output. ### ico ICO file muxer. Microsoft's icon file format (ICO) has some strict limitations that should be noted: • Size cannot exceed 256 pixels in any dimension • Only BMP and PNG images can be stored • If a BMP image is used, it must be one of the following pixel formats: BMP Bit Depth FFmpeg Pixel Format 1bit pal8 4bit pal8 8bit pal8 16bit rgb555le 24bit bgr24 32bit bgra • If a BMP image is used, it must use the BITMAPINFOHEADER DIB header • If a PNG image is used, it must use the rgba pixel format ### image2 Image file muxer. The image file muxer writes video frames to image files. The output filenames are specified by a pattern, which can be used to produce sequentially numbered series of files. The pattern may contain the string "%d" or "%0_N_d", this string specifies the position of the characters representing a numbering in the filenames. If the form "%0_N_d" 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 "%0_N_d", the first filename of the file list specified will contain the number 1, all the following numbers will be sequential. The pattern may contain a suffix which is used to automatically determine the format of the image files to write. For example the pattern "img-%03d.bmp" will specify a sequence of filenames of the form _img-001.bmp_, _img-002.bmp_, ..., _img-010.bmp_, etc. The pattern "img%%-%d.jpg" will specify a sequence of filenames of the form _img%-1.jpg_, _img%-2.jpg_, ..., _img%-10.jpg_, etc. The image muxer supports the .Y.U.V image file format. This format is special in that that each image frame consists of three files, for each of the YUV420P components. To read or write this image file format, specify the name of the '.Y' file. The muxer will automatically open the '.U' and '.V' files as required. _Options_ ### frame_pts If set to 1, expand the filename with pts from pkt->pts. Default value is 0. ### start_number Start the sequence from the specified number. Default value is 1. ### update If set to 1, the filename will always be interpreted as just a filename, not a pattern, and the corresponding file will be continuously overwritten with new images. Default value is 0. ### strftime If set to 1, expand the filename with date and time information from strftime(). Default value is 0. ### atomic_writing Write output to a temporary file, which is renamed to target filename once writing is completed. Default is disabled. **protocol_opts _**options_list_ Set protocol options as a :-separated list of key=value parameters. Values containing the ":" special character must be escaped. _Examples_ The following example shows how to use **ffmpeg **for creating a sequence of files _img-001.jpeg_, _img-002.jpeg_, ..., taking one image every second from the input video: ffmpeg -i in.avi -vsync cfr -r 1 -f image2 'img-%03d.jpeg' Note that with **ffmpeg**, if the format is not specified with the "-f" option and the output filename specifies an image file format, the image2 muxer is automatically selected, so the previous command can be written as: ffmpeg -i in.avi -vsync cfr -r 1 'img-%03d.jpeg' Note also that the pattern must not necessarily contain "%d" or "%0_N_d", for example to create a single image file _img.jpeg_ from the start of the input video you can employ the command: ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg The **strftime **option allows you to expand the filename with date and time information. Check the documentation of the strftime() function for the syntax. For example to generate image files from the strftime() "%Y-%m-%d_%H-%M-%S" pattern, the following **ffmpeg **command can be used: ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg" You can set the file name with current frame's PTS: ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg A more complex example is to publish contents of your desktop directly to a WebDAV server every second: ffmpeg -f x11grab -framerate 1 -i :0.0 -q:v 6 -update 1 -protocol_opts method=PUT ### matroska Matroska container muxer. This muxer implements the matroska and webm container specs. _Metadata_ The recognized metadata settings in this muxer are: ### title Set title name provided to a single track. This gets mapped to the FileDescription element for a stream written as attachment. ### language Specify the language of the track in the Matroska languages form. The language can be either the 3 letters bibliographic ISO-639-2 (ISO 639-2/B) form (like "fre" for French), or a language code mixed with a country code for specialities in languages (like "fre-ca" for Canadian French). ### stereo_mode Set stereo 3D video layout of two views in a single video track. The following values are recognized: **mono** video is not stereo **left_right** Both views are arranged side by side, Left-eye view is on the left **bottom_top** Both views are arranged in top-bottom orientation, Left-eye view is at bottom **top_bottom** Both views are arranged in top-bottom orientation, Left-eye view is on top **checkerboard_rl** Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first **checkerboard_lr** Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first **row_interleaved_rl** Each view is constituted by a row based interleaving, Right-eye view is first row **row_interleaved_lr** Each view is constituted by a row based interleaving, Left-eye view is first row **col_interleaved_rl** Both views are arranged in a column based interleaving manner, Right-eye view is first column **col_interleaved_lr** Both views are arranged in a column based interleaving manner, Left-eye view is first column **anaglyph_cyan_red** All frames are in anaglyph format viewable through red-cyan filters **right_left** Both views are arranged side by side, Right-eye view is on the left **anaglyph_green_magenta** All frames are in anaglyph format viewable through green-magenta filters **block_lr** Both eyes laced in one Block, Left-eye view is first **block_rl** Both eyes laced in one Block, Right-eye view is first For example a 3D WebM clip can be created using the following command line: ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm _Options_ This muxer supports the following options: ### reserve_index_space By default, this muxer writes the index for seeking (called cues in Matroska terms) at the end of the file, because it cannot know in advance how much space to leave for the index at the beginning of the file. However for some use cases -- e.g. streaming where seeking is possible but slow -- it is useful to put the index at the beginning of the file. If this option is set to a non-zero value, the muxer will reserve a given amount of space in the file header and then try to write the cues there when the muxing finishes. If the 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. ### cues_to_front 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. ### default_mode This option controls how the FlagDefault of the output tracks will be set. It influences which tracks players should play by default. The default mode is **passthrough**. **infer** 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. **infer_no_subs** 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. **passthrough** 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. ### flipped_raw_rgb If set to true, store positive height for raw RGB bitmaps, which indicates bitmap is stored bottom-up. Note that this option does not flip the bitmap which has to be done manually beforehand, e.g. by using the vflip filter. Default is _false_ and indicates bitmap is stored top down. ### md5 MD5 testing format. This is a variant of the **hash **muxer. Unlike that muxer, it defaults to using the MD5 hash function. _Examples_ To compute the MD5 hash of the input converted to raw audio and video, and store it in the file _out.md5_: ffmpeg -i INPUT -f md5 out.md5 You can print the MD5 to stdout with the command: ffmpeg -i INPUT -f md5 - See also the **hash **and **framemd5 **muxers. ### mov, mp4, ismv MOV/MP4/ISMV (Smooth Streaming) muxer. The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4 file has all the metadata about all packets stored in one location (written at the end of the file, it can be moved to the start for better playback by adding "+faststart" to the "-movflags", or using the **qt-** **faststart **tool). A fragmented file consists of a number of fragments, where packets and metadata about these packets are stored together. Writing a fragmented file has the advantage that the file is decodable even if the writing is interrupted (while a normal MOV/MP4 is undecodable if it is not properly finished), and it requires less memory when writing very long files (since writing normal MOV/MP4 files stores info about every single packet in memory until the file is closed). The downside is that it is less compatible with other applications. Fragmentation is enabled by setting one of the options that define how to cut the file into fragments: "-frag_duration", "-frag_size", "-min_frag_duration", "-movflags +frag_keyframe" and "-movflags +frag_custom". If more than one condition is specified, fragments are cut when one of the specified conditions is fulfilled. The exception to this is "-min_frag_duration", which has to be fulfilled for any of the other conditions to apply. _Options_ **frag_duration _**duration_ Create fragments that are _duration_ microseconds long. **frag_size _**size_ Create fragments that contain up to _size_ bytes of payload data. **min_frag_duration _**duration_ Don't create fragments that are shorter than _duration_ microseconds long. **movflags _**flags_ Set various muxing switches. The following flags can be used: **frag_keyframe** Start a new fragment at each video keyframe. **frag_custom** Allow the caller to manually choose when to cut fragments, by calling "av_write_frame(ctx, NULL)" to write a fragment with the packets written so far. (This is only useful with other applications integrating libavformat, not from **ffmpeg**.) **empty_moov** Write an initial moov atom directly at the start of the file, without describing any samples in it. Generally, an mdat/moov pair is written at the start of the file, as a normal MOV/MP4 file, containing only a short portion of the file. With this option set, there is no initial mdat atom, and the moov atom only describes the tracks but has a zero duration. This option is implicitly set when writing ismv (Smooth Streaming) files. **separate_moof** Write a separate moof (movie fragment) atom for each track. Normally, packets for all tracks are written in a moof atom (which is slightly more efficient), but with this option set, the muxer writes one moof/mdat pair for each track, making it easier to separate tracks. This option is implicitly set when writing ismv (Smooth Streaming) files. **skip_sidx** Skip writing of sidx atom. When bitrate overhead due to sidx atom is high, this option could be used for cases where sidx atom is not mandatory. When global_sidx flag is enabled, this option will be ignored. **faststart** Run a second pass moving the index (moov atom) to the beginning of the file. This operation can take a while, and will not work in various situations such as fragmented output, thus it is not enabled by default. **rtphint** Add RTP hinting tracks to the output file. **disable_chpl** Disable Nero chapter markers (chpl atom). Normally, both Nero chapters and a QuickTime chapter track are written to the file. With this option set, only the QuickTime chapter track will be written. Nero chapters can cause failures when the file is reprocessed with certain tagging programs, like mp3Tag 2.61a and iTunes 11.3, most likely other versions are affected as well. **omit_tfhd_offset** Do not write any absolute base_data_offset in tfhd atoms. This avoids tying fragments to absolute byte positions in the file/streams. **default_base_moof** Similarly to the omit_tfhd_offset, this flag avoids writing the absolute base_data_offset field in tfhd atoms, but does so by using the new default-base-is- moof flag instead. This flag is new from 14496-12:2012. This may make the fragments easier to parse in certain circumstances (avoiding basing track fragment location calculations on the implicit end of the previous track fragment). **negative_cts_offsets** 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. **moov_size _**bytes_ Reserves space for the moov atom at the beginning of the file instead of placing the moov atom at the end. If the space reserved is insufficient, muxing will fail. ### write_tmcd Specify "on" to force writing a timecode track, "off" to disable it and "auto" to write a timecode track only for mov and mp4 output (default). **write_btrt _**bool_ 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_prft Write producer time reference box (PRFT) with a specified time source for the NTP field in the PRFT box. Set value as **wallclock **to specify timesource as wallclock time and **pts** to specify timesource as input packets' PTS values. Setting value to **pts **is applicable only for a live encoding use case, where PTS values are set as as wallclock time at the source. For example, an encoding use case with decklink capture source where **video_pts **and **audio_pts **are set to **abs_wallclock**. **empty_hdlr_name _**bool_ Enable to skip writing the name inside a "hdlr" box. Default is "false". **movie_timescale _**scale_ Set the timescale written in the movie header box ("mvhd"). Range is 1 to INT_MAX. Default is 1000. **video_track_timescale _**scale_ 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. _Example_ Smooth Streaming content can be pushed in real time to a publishing point on IIS with this muxer. Example: ffmpeg -re <> -movflags isml+frag_keyframe -f ismv (Encoder1) ### mp3 The MP3 muxer writes a raw MP3 stream with the following optional features: • An ID3v2 metadata header at the beginning (enabled by default). Versions 2.3 and 2.4 are supported, the "id3v2_version" private option controls which one is used (3 or 4). Setting "id3v2_version" to 0 disables the ID3v2 header completely. The muxer supports writing attached pictures (APIC frames) to the ID3v2 header. The pictures are supplied to the muxer in form of a video stream with a single packet. There can be any number of those streams, each will correspond to a single APIC frame. The stream metadata tags _title_ and _comment_ map to APIC _description_ and _picture_ _type_ respectively. See <****> for allowed picture types. Note that the APIC frames must be written at the beginning, so the muxer will buffer the audio frames until it gets all the pictures. It is therefore advised to provide the pictures as soon as possible to avoid excessive buffering. • A Xing/LAME frame right after the ID3v2 header (if present). It is enabled by default, but will be written only if the output is seekable. The "write_xing" private option can be used to disable it. The frame contains various information that may be useful to the decoder, like the audio duration or encoder delay. • A legacy ID3v1 tag at the end of the file (disabled by default). It may be enabled with the "write_id3v1" private option, but as its capabilities are very limited, its usage is not recommended. Examples: Write an mp3 with an ID3v2.3 header and an ID3v1 footer: ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 To attach a picture to an mp3 file select both the audio and the picture stream with "map": ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1 -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3 Write a "clean" MP3 without any extra features: ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3 ### mpegts MPEG transport stream muxer. This muxer implements ISO 13818-1 and part of ETSI EN 300 468. The recognized metadata settings in mpegts muxer are "service_provider" and "service_name". If they are not set the default for "service_provider" is **FFmpeg **and the default for "service_name" is **Service01**. _Options_ The muxer options are: **mpegts_transport_stream_id _**integer_ Set the **transport_stream_id**. This identifies a transponder in DVB. Default is 0x0001. **mpegts_original_network_id _**integer_ 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. **mpegts_service_id _**integer_ Set the **service_id**, also known as program in DVB. Default is 0x0001. **mpegts_service_type _**integer_ Set the program **service_type**. Default is "digital_tv". Accepts the following options: **hex_value** Any hexadecimal value between 0x01 and 0xff as defined in ETSI 300 468. **digital_tv** Digital TV service. **digital_radio** Digital Radio service. **teletext** Teletext service. **advanced_codec_digital_radio** Advanced Codec Digital Radio service. **mpeg2_digital_hdtv** MPEG2 Digital HDTV service. **advanced_codec_digital_sdtv** Advanced Codec Digital SDTV service. **advanced_codec_digital_hdtv** Advanced Codec Digital HDTV service. **mpegts_pmt_start_pid _**integer_ 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. **mpegts_start_pid _**integer_ 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. **mpegts_m2ts_mode _**boolean_ Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode. **muxrate _**integer_ Set a constant muxrate. Default is VBR. **pes_payload_size _**integer_ Set minimum PES packet payload in bytes. Default is 2930. **mpegts_flags _**flags_ Set mpegts flags. Accepts the following options: **resend_headers** Reemit PAT/PMT before writing the next packet. **latm** Use LATM packetization for AAC. **pat_pmt_at_frames** Reemit PAT and PMT at each video frame. **system_b** Conform to System B (DVB) instead of System A (ATSC). **initial_discontinuity** Mark the initial packet of each stream as discontinuity. **nit **Emit NIT table. **omit_rai** Disable writing of random access indicator. **mpegts_copyts _**boolean_ 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_video_pes_length _**boolean_ Omit the PES packet length for video packets. Default is 1 (true). **pcr_period _**integer_ 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. **pat_period _**duration_ Maximum time in seconds between PAT/PMT tables. Default is 0.1. **sdt_period _**duration_ Maximum time in seconds between SDT tables. Default is 0.5. **nit_period _**duration_ Maximum time in seconds between NIT tables. Default is 0.5. **tables_version _**integer_ 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, mxf_d10, mxf_opatom MXF muxer. _Options_ The muxer options are: **store_user_comments _**bool_ Set if user comments should be stored if available or never. IRT D-10 does not allow user comments. The default is thus to write them for mxf and mxf_opatom but not for mxf_d10 ### null Null muxer. This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes. For example to benchmark decoding with **ffmpeg **you can use the command: ffmpeg -benchmark -i INPUT -f null out.null Note that the above command does not read or write the _out.null_ file, but specifying the output file is required by the **ffmpeg **syntax. Alternatively you can write the command as: ffmpeg -benchmark -i INPUT -f null - ### nut ### -syncpoints _ Change the syncpoint usage in nut: _default_ **use the normal low-overhead seeking aids.** _none_ **do not use the syncpoints at all, reducing the overhead but making the stream non-** **seekable;** Use of this option is not recommended, as the resulting files are very damage sensitive and seeking is not possible. Also in general the overhead from syncpoints is negligible. Note, -C 0 can be used to disable all growing data tables, allowing to mux endless streams with limited memory and without these disadvantages. _timestamped_ **extend the syncpoint with a wallclock field.** The _none_ and _timestamped_ flags are experimental. ### -write_index _ Write index at the end, the default is to write an index. ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor ### ogg Ogg container muxer. ### -page_duration _ Preferred page duration, in microseconds. The muxer will attempt to create pages that are approximately _duration_ microseconds long. This allows the user to compromise between seek granularity and container overhead. The default is 1 second. A value of 0 will fill all segments, making pages as large as possible. A value of 1 will effectively use 1 packet- per-page in most situations, giving a small seek granularity at the cost of additional container overhead. ### -serial_offset _ 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. ### raw muxers Raw muxers 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. _ac3_ Dolby Digital, also known as AC-3, audio. _adx_ 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_ aptX (Audio Processing Technology for Bluetooth) audio. _aptx_hd_ aptX HD (Audio Processing Technology for Bluetooth) audio. Extensions: aptxhd _avs2_ AVS2-P2/IEEE1857.4 video. Extensions: avs, avs2 _cavsvideo_ Chinese AVS (Audio Video Standard) video. Extensions: cavs _codec2raw_ Codec 2 audio. No extension is registered so format name has to be supplied e.g. with the ffmpeg CLI tool "-f codec2raw". _data_ Data 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". _dirac_ BBC Dirac video. The Dirac Pro codec is a subset and is standardized as SMPTE VC-2. Extensions: drc, vc2 _dnxhd_ Avid DNxHD video. It is standardized as SMPTE VC-3. Accepts DNxHR streams. Extensions: dnxhd, dnxhr _dts_ DTS Coherent Acoustics (DCA) audio. _eac3_ Dolby Digital Plus, also known as Enhanced AC-3, audio. _evc_ MPEG-5 Essential Video Coding (EVC) / EVC / MPEG-5 Part 1 EVC video. Extensions: evc _g722_ ITU-T G.722 audio. _g723_1_ ITU-T G.723.1 audio. Extensions: tco, rco _g726_ 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". _g726le_ 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". _gsm_ Global System for Mobile Communications audio. _h261_ ITU-T H.261 video. _h263_ ITU-T H.263 / H.263-1996, H.263+ / H.263-1998 / H.263 version 2 video. _h264_ 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. Extensions: h264, 264 _hevc_ 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. Extensions: hevc, h265, 265 _m4v_ MPEG-4 Part 2 video. _mjpeg_ Motion JPEG video. Extensions: mjpg, mjpeg _mlp_ Meridian Lossless Packing, also known as Packed PCM, audio. _mp2_ MPEG-1 Audio Layer II audio. Extensions: mp2, m2a, mpa _mpeg1video_ MPEG-1 Part 2 video. Extensions: mpg, mpeg, m1v _mpeg2video_ ITU-T H.262 / MPEG-2 Part 2 video. Extensions: m2v _obu_ AV1 low overhead Open Bitstream Units muxer. Temporal delimiter OBUs will be inserted in all temporal units of the stream. _rawvideo_ Raw uncompressed video. Extensions: yuv, rgb _sbc_ Bluetooth SIG low-complexity subband codec audio. Extensions: sbc, msbc _truehd_ Dolby TrueHD audio. Extensions: thd _vc1_ SMPTE 421M / VC-1 video. ### segment, stream_segment, ssegment Basic stream segmenter. This muxer outputs streams to a number of separate files of nearly fixed duration. Output filename pattern can be set in a fashion similar to **image2**, or by using a "strftime" template if the **strftime **option is enabled. "stream_segment" is a variant of the muxer used to write to streaming output formats, i.e. which do not require global headers, and is recommended for outputting e.g. to MPEG transport stream segments. "ssegment" is a shorter alias for "stream_segment". Every segment starts with a keyframe of the selected reference stream, which is set through the **reference_stream **option. Note that if you want accurate splitting for a video file, you need to make the input key frames correspond to the exact splitting times expected by the segmenter, or the segment muxer will start the new segment with the key frame found next after the specified start time. The segment muxer works best with a single constant frame rate video. Optionally it can generate a list of the created segments, by setting the option _segment_list_. The list type is specified by the _segment_list_type_ option. The entry filenames in the segment list are set by default to the basename of the corresponding segment files. See also the **hls **muxer, which provides a more specific implementation for HLS segmentation. _Options_ The segment muxer supports the following options: **increment_tc _**1|0_ if set to 1, increment timecode between each segment If this is selected, the input need to have a timecode in the first video stream. Default value is 0. **reference_stream _**specifier_ Set the reference stream, as specified by the string _specifier_. If _specifier_ is set to "auto", the reference is chosen automatically. Otherwise it must be a stream specifier (see the ``Stream specifiers'' chapter in the ffmpeg manual) which specifies the reference stream. The default value is "auto". **segment_format _**format_ Override the inner container format, by default it is guessed by the filename extension. **segment_format_options _**options_list_ Set output format options using a :-separated list of key=value parameters. Values containing the ":" special character must be escaped. **segment_list _**name_ Generate also a listfile named _name_. If not specified no listfile is generated. **segment_list_flags _**flags_ Set flags affecting the segment list generation. It currently supports the following flags: **cache** Allow caching (only affects M3U8 list files). **live** Allow live-friendly file generation. **segment_list_size _**size_ Update the list file so that it contains at most _size_ segments. If 0 the list file will contain all the segments. Default value is 0. **segment_list_entry_prefix _**prefix_ Prepend _prefix_ to each entry. Useful to generate absolute paths. By default no prefix is applied. **segment_list_type _**type_ Select the listing format. The following values are recognized: **flat** Generate a flat list for the created segments, one segment per line. **csv, ext** Generate a list for the created segments, one segment per line, each line matching the format (comma-separated values): ,, _segment_filename_ is the name of the output file generated by the muxer according to the provided pattern. CSV escaping (according to RFC4180) is applied if required. _segment_start_time_ and _segment_end_time_ specify the segment start and end time expressed in seconds. A list file with the suffix ".csv" or ".ext" will auto-select this format. **ext **is deprecated in favor or **csv**. **ffconcat** Generate an ffconcat file for the created segments. The resulting file can be read using the FFmpeg **concat **demuxer. A list file with the suffix ".ffcat" or ".ffconcat" will auto-select this format. **m3u8** Generate an extended M3U8 file, version 3, compliant with <****>. A list file with the suffix ".m3u8" will auto-select this format. If not specified the type is guessed from the list file name suffix. **segment_time _**time_ Set segment duration to _time_, the value must be a duration specification. Default value is "2". See also the **segment_times **option. Note that splitting may not be accurate, unless you force the reference stream key-frames at the given time. See the introductory notice and the examples below. **min_seg_duration _**time_ 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". **segment_atclocktime _**1|0_ If set to "1" split at regular clock time intervals starting from 00:00 o'clock. The _time_ value specified in **segment_time **is used for setting the length of the splitting interval. For example with **segment_time **set to "900" this makes it possible to create files at 12:00 o'clock, 12:15, 12:30, etc. Default value is "0". **segment_clocktime_offset _**duration_ Delay the segment splitting times with the specified duration when using **segment_atclocktime**. For example with **segment_time **set to "900" and **segment_clocktime_offset **set to "300" this makes it possible to create files at 12:05, 12:20, 12:35, etc. Default value is "0". **segment_clocktime_wrap_duration _**duration_ Force the segmenter to only start a new segment if a packet reaches the muxer within the specified duration after the segmenting clock time. This way you can make the segmenter more resilient to backward local time jumps, such as leap seconds or transition to standard time from daylight savings time. Default is the maximum possible duration which means starting a new segment regardless of the elapsed time since the last clock time. **segment_time_delta _**delta_ Specify the accuracy time when selecting the start time for a segment, expressed as a duration specification. Default value is "0". When delta is specified a key-frame will start a new segment if its PTS satisfies the relation: PTS >= start_time - time_delta This option is useful when splitting video content, which is always split at GOP boundaries, in case a key frame is found just before the specified split time. In particular may be used in combination with the _ffmpeg_ option _force_key_frames_. The key frame times specified by _force_key_frames_ may not be set accurately because of rounding issues, with the consequence that a key frame time may result set just before the specified time. For constant frame rate videos a value of 1/(2*_frame_rate_) should address the worst case mismatch between the specified time and the time set by _force_key_frames_. **segment_times _**times_ Specify a list of split points. _times_ contains a list of comma separated duration specifications, in increasing order. See also the **segment_time **option. **segment_frames _**frames_ Specify a list of split video frame numbers. _frames_ contains a list of comma separated integer numbers, in increasing order. This option specifies to start a new segment whenever a reference stream key frame is found and the sequential number (starting from 0) of the frame is greater or equal to the next value in the list. **segment_wrap _**limit_ Wrap around segment index once it reaches _limit_. **segment_start_number _**number_ Set the sequence number of the first segment. Defaults to 0. **strftime _**1|0_ Use the "strftime" function to define the name of the new segments to write. If this is selected, the output segment name must contain a "strftime" function template. Default value is 0. **break_non_keyframes _**1|0_ If enabled, allow segments to start on frames other than keyframes. This improves behavior on some players when the time between keyframes is inconsistent, but may make things worse on others, and can cause some oddities during seeking. Defaults to 0. **reset_timestamps _**1|0_ Reset timestamps at the 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. **initial_offset _**offset_ Specify timestamp offset to apply to the output packet timestamps. The argument must be a time duration specification, and defaults to 0. **write_empty_segments _**1|0_ If enabled, write an empty segment if there are no packets during the period a segment would usually span. Otherwise, the segment will be filled with the next packet written. Defaults to 0. 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 ### smoothstreaming Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving with conventional web server. ### window_size Specify the number of fragments kept in the manifest. Default 0 (keep all). ### extra_window_size Specify the number of fragments kept outside of the manifest before removing from disk. Default 5. ### lookahead_count Specify the number of lookahead fragments. Default 2. ### min_frag_duration Specify the minimum fragment duration (in microseconds). Default 5000000. ### remove_at_exit Specify whether to remove all fragments when finished. Default 0 (do not remove). ### streamhash 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. ### tee The tee muxer can be used to write the same data to several outputs, such as files or streams. It can be used, for example, to stream a video over a network and save it to disk at the same time. It is different from specifying several outputs to the **ffmpeg **command-line tool. With the tee muxer, the audio and video data will be encoded only once. With conventional multiple outputs, multiple encoding operations in parallel are initiated, which can be a very expensive process. The tee muxer is not useful when using the libavformat API directly because it is then possible to feed the same packets to several muxers directly. Since the tee muxer does not represent any particular output format, ffmpeg cannot auto- select output streams. So all streams intended for output must be specified using "-map". See the examples below. Some encoders may need different options depending on the output format; the auto-detection of this can not work with the tee muxer, so they need to be explicitly specified. The main example is the **global_header **flag. The slave outputs are specified in the file name given to the muxer, separated by '|'. If any of the slave name contains the '|' separator, leading or trailing spaces or any special character, those must be escaped (see **the "Quoting and escaping" section in the** [**ffmpeg-utils(1)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown) manual**). _Options_ **use_fifo _**bool_ If set to 1, slave outputs will be processed in separate threads using the **fifo **muxer. This allows to compensate for different speed/latency/reliability of outputs and setup transparent recovery. By default this feature is turned off. ### fifo_options Options to pass to fifo pseudo-muxer instances. See **fifo**. Muxer options can be specified for each slave by prepending them as a list of _key_=_value_ pairs separated by ':', between square brackets. If the options values contain a special character or the ':' separator, they must be escaped; note that this is a second level escaping. The following special options are also recognized: **f **Specify the format name. Required if it cannot be guessed from the output URL. **bsfs[/_**spec_**]** Specify a list of bitstream filters to apply to the specified output. It is possible to specify to which streams a given bitstream filter applies, by appending a stream specifier to the option separated by "/". _spec_ must be a stream specifier (see **Format stream specifiers**). If the stream specifier is not specified, the bitstream filters will be applied to all streams in the output. This will cause that output operation to fail if the output contains streams to which the bitstream filter cannot be applied e.g. "h264_mp4toannexb" being applied to an output containing an audio stream. Options for a bitstream filter must be specified in the form of "opt=value". Several bitstream filters can be specified, separated by ",". **use_fifo _**bool_ This allows to override tee muxer use_fifo option for individual slave muxer. ### fifo_options This allows to override tee muxer fifo_options for individual slave muxer. See **fifo**. ### select Select the streams that should be mapped to the slave output, specified by a stream specifier. If not specified, this defaults to all the mapped streams. This will cause that output operation to fail if the output format does not accept all mapped streams. You may use multiple stream specifiers separated by commas (",") e.g.: "a:0,v" ### onfail Specify behaviour on output failure. This can be set to either "abort" (which is default) or "ignore". "abort" will cause whole process to fail in case of failure on this slave output. "ignore" will ignore failure on this output, so other outputs will continue without being affected. _Examples_ • Encode something and both archive it in a WebM file and stream it as MPEG-TS over UDP: ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/" • As above, but continue streaming even if output to local file fails (for example local drive fills up): ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a "[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/" • Use **ffmpeg **to encode the input, and send the output to three different destinations. The "dump_extra" bitstream filter is used to add extradata information to all the output video keyframes packets, as requested by the MPEG-TS format. The select option is applied to _out.aac_ in order to make it contain only audio packets. ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac" • As above, but select only stream "a:1" for the audio output. Note that a second level escaping must be performed, as ":" is a special character used to separate options. ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac" ### webm_chunk WebM Live Chunk Muxer. This muxer writes out WebM headers and chunks as separate files which can be consumed by clients that support WebM Live streams via DASH. _Options_ This muxer supports the following options: ### chunk_start_index Index of the first chunk (defaults to 0). ### header Filename of the header where the initialization data will be written. ### audio_chunk_duration Duration of each audio chunk in milliseconds (defaults to 5000). _Example_ ffmpeg -f v4l2 -i /dev/video0 \ -f alsa -i hw:0 \ -map 0:0 \ -c:v libvpx-vp9 \ -s 640x360 -keyint_min 30 -g 30 \ -f webm_chunk \ -header webm_live_video_360.hdr \ -chunk_start_index 1 \ webm_live_video_360_%d.chk \ -map 1:0 \ -c:a libvorbis \ -b:a 128k \ -f webm_chunk \ -header webm_live_audio_128.hdr \ -chunk_start_index 1 \ -audio_chunk_duration 1000 \ webm_live_audio_128_%d.chk ### webm_dash_manifest WebM DASH Manifest muxer. This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML. It also supports manifest generation for DASH live streams. For more information see: • WebM DASH Specification: <****> • ISO DASH Specification: <****> _Options_ This muxer supports the following options: ### adaptation_sets This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding audio and video streams. Any number of adaptation sets can be added using this option. ### live Set this to 1 to create a live stream DASH Manifest. Default: 0. ### chunk_start_index Start index of the first chunk. This will go in the **startNumber **attribute of the **SegmentTemplate **element in the manifest. Default: 0. ### chunk_duration_ms Duration of each chunk in milliseconds. This will go in the **duration **attribute of the **SegmentTemplate **element in the manifest. Default: 1000. ### utc_timing_url URL of the page that will return the UTC timestamp in ISO format. This will go in the **value **attribute of the **UTCTiming **element in the manifest. Default: None. ### time_shift_buffer_depth Smallest time (in seconds) shifting buffer for which any Representation is guaranteed to be available. This will go in the **timeShiftBufferDepth **attribute of the **MPD **element. Default: 60. ### minimum_update_period Minimum update period (in seconds) of the manifest. This will go in the **minimumUpdatePeriod **attribute of the **MPD **element. Default: 0. _Example_ ffmpeg -f webm_dash_manifest -i video1.webm \ -f webm_dash_manifest -i video2.webm \ -f webm_dash_manifest -i audio1.webm \ -f webm_dash_manifest -i audio2.webm \ -map 0 -map 1 -map 2 -map 3 \ -c copy \ -f webm_dash_manifest \ -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \ manifest.xml ## METADATA FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded INI-like text file and then load it back using the metadata muxer/demuxer. The file format is as follows: 1. A file consists of a header and a number of metadata tags divided into sections, each on its own line. 2. The header is a **;FFMETADATA **string, followed by a version number (now 1). 3. Metadata tags are of the form **key=value** 4. Immediately after header follows global metadata 5. After global metadata there may be sections with per-stream/per-chapter metadata. 6. A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in brackets (**[**, **]**) and ends with next section or end of file. 7. At the beginning of a chapter section there may be an optional timebase to be used for start/end values. It must be in form **TIMEBASE=_**num_**/_**den_, where _num_ and _den_ are integers. If the timebase is missing then start/end times are assumed to be in nanoseconds. Next a chapter section must contain chapter start and end times in form **START=_**num_, **END=_**num_, where _num_ is a positive integer. 8. Empty lines and lines starting with **; **or **# **are ignored. 9. Metadata keys or values containing special characters (**=**, **;**, **#**, **\ **and a newline) must be escaped with a backslash **\**. 10. Note that whitespace in metadata (e.g. **foo = bar**) is considered to be a part of the tag (in the example above key is **foo **, value is **bar**). A ffmetadata file might look like this: ;FFMETADATA1 title=bike\\shed ;this is a comment artist=FFmpeg troll team [CHAPTER] TIMEBASE=1/1000 START=0 #chapter ends at 0:01:00 END=60000 title=chapter \#1 [STREAM] title=multi\ line By using the ffmetadata muxer and demuxer it is possible to extract metadata from an input file to an ffmetadata file, and then transcode the file into an output file with the edited ffmetadata file. Extracting an ffmetadata file with _ffmpeg_ goes as follows: ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE Reinserting edited metadata information from the FFMETADATAFILE file can be done as: ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT ## PROTOCOL OPTIONS The libavformat library provides some generic global options, which can be set on all the protocols. In addition each protocol may support so-called private options, which are specific for that component. Options may be set by specifying -_option_ _value_ in the FFmpeg tools, or by setting the value explicitly in the "AVFormatContext" options or using the _libavutil/opt.h_ API for programmatic use. The list of supported options follows: **protocol_whitelist _**list_ **(_**input_**)** Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols prefixed by "-" are disabled. All protocols are allowed by default but protocols used by an another protocol (nested protocols) are restricted to a per protocol subset. ## PROTOCOLS Protocols are configured elements in FFmpeg that enable access to resources that require specific protocols. When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list all available ones using the configure option "--list-protocols". You can disable all the protocols using the configure option "--disable-protocols", and selectively enable a protocol using the option "--enable-protocol=_PROTOCOL_", or you can disable a particular protocol using the option "--disable-protocol=_PROTOCOL_". The option "-protocols" of the ff* tools will display the list of supported protocols. All protocols accept the following options: ### rw_timeout Maximum time to wait for (network) read/write operations to complete, in microseconds. A description of the currently available protocols follows. ### amqp Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based publish-subscribe communication protocol. FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A separate AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ. After starting the broker, an FFmpeg client may stream data to the broker using the command: ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost] Where hostname and port (default is 5672) is the address of the broker. The client may also set a user/password for authentication. The default for both fields is "guest". Name of virtual host on broker can be set with vhost. The default value is "/". Muliple subscribers may stream from the broker using the command: ffplay amqp://[[user]:[password]@]hostname[:port][/vhost] In RabbitMQ all data published to the broker flows through a specific exchange, and each subscribing client has an assigned queue/buffer. When a packet arrives at an exchange, it may be copied to a client's queue depending on the exchange and routing_key fields. The following options are supported: ### exchange Sets the exchange to use on the broker. RabbitMQ has several predefined exchanges: "amq.direct" is the default exchange, where the publisher and subscriber must have a matching routing_key; "amq.fanout" is the same as a broadcast operation (i.e. the data is forwarded to all queues on the fanout exchange independent of the routing_key); and "amq.topic" is similar to "amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ documentation). ### routing_key Sets the routing key. The default value is "amqp". The routing key is used on the "amq.direct" and "amq.topic" exchanges to decide whether packets are written to the queue of a subscriber. ### pkt_size Maximum size of each packet sent/received to the broker. Default is 131072. Minimum is 4096 and max is any large value (representable by an int). When receiving packets, this sets an internal buffer size in FFmpeg. It should be equal to or greater than the size of the published packets to the broker. Otherwise the received message may be truncated causing decoding errors. ### connection_timeout The timeout in seconds during the initial connection to the broker. The default value is rw_timeout, or 5 seconds if rw_timeout is not set. **delivery_mode _**mode_ Sets the delivery mode of each message sent to broker. The following values are accepted: **persistent** Delivery mode set to "persistent" (2). This is the default value. Messages may be written to the broker's disk depending on its setup. **non-persistent** Delivery mode set to "non-persistent" (1). Messages will stay in broker's memory unless the broker is under memory pressure. ### async Asynchronous data filling wrapper for input stream. Fill data in a background thread, to decouple I/O operation from demux thread. async: async: async:cache: ### bluray Read BluRay playlist. The accepted options are: ### angle BluRay angle ### chapter Start chapter (1...N) ### playlist Playlist to read (BDMV/PLAYLIST/?????.mpls) Examples: Read longest playlist from BluRay mounted to /mnt/bluray: bluray:/mnt/bluray Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray ### cache Caching wrapper for input stream. Cache the input stream to temporary file. It brings seeking capability to live streams. The accepted options are: ### read_ahead_limit Amount in bytes that may be read ahead when seeking isn't supported. Range is -1 to INT_MAX. -1 for unlimited. Default is 65536. URL Syntax is cache: ### concat Physical concatenation protocol. Read and seek from many resources in sequence as if they were a unique resource. A URL accepted by this protocol has the syntax: concat:||...| where _URL1_, _URL2_, ..., _URLN_ are the urls of the resource to be concatenated, each one possibly specifying a distinct protocol. For example to read a sequence of files _split1.mpeg_, _split2.mpeg_, _split3.mpeg_ with **ffplay **use the command: ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg Note that you may need to escape the character "|" which is special for many shells. ### concatf Physical concatenation protocol using a line break delimited list of resources. Read and seek from many resources in sequence as if they were a unique resource. A URL accepted by this protocol has the syntax: concatf: where _URL_ is the url containing a line break delimited list of resources to be concatenated, each one possibly specifying a distinct protocol. Special characters must be escaped with backslash or single quotes. See **the "Quoting and escaping" section in the [ffmpeg-utils(1)](https://www.chedong.com/phpMan.php/man/ffmpeg-utils/1/markdown)** **manual**. For example to read a sequence of files _split1.mpeg_, _split2.mpeg_, _split3.mpeg_ listed in separate lines within a file _split.txt_ with **ffplay **use the command: ffplay concatf:split.txt Where _split.txt_ contains the lines: split1.mpeg split2.mpeg split3.mpeg ### crypto AES-encrypted stream reading protocol. The accepted options are: **key **Set the AES decryption key binary block from given hexadecimal representation. **iv **Set the AES decryption initialization vector binary block from given hexadecimal representation. Accepted URL formats: crypto: crypto+ ### data Data in-line in the URI. See <****>. For example, to convert a GIF file given inline with **ffmpeg**: ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png **fd** File descriptor access protocol. The accepted syntax is: fd: -fd If **fd **is not specified, by default the stdout file descriptor will be used for writing, stdin for reading. Unlike the pipe protocol, fd protocol has seek support if it corresponding to a regular file. fd protocol doesn't support pass file descriptor via URL for security. This protocol accepts the following options: ### blocksize Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable if data transmission is slow. **fd **Set file descriptor. ### file File access protocol. Read from or write to a file. A file URL can have the form: file: where _filename_ is the path of the file to read. An URL that does not have a protocol prefix will be assumed to be a file URL. Depending on the build, an URL that looks like a Windows path with the drive letter at the beginning will also be assumed to be a file URL (usually not the case in builds for unix-like systems). For example to read from a file _input.mpeg_ with **ffmpeg **use the command: ffmpeg -i file:input.mpeg output.mpeg This protocol accepts the following options: ### truncate Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1. ### blocksize Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable for files on slow medium. ### follow If set to 1, the protocol will retry reading at the end of the file, allowing reading files that still are being written. In order for this to terminate, you either need to use the rw_timeout option, or use the interrupt callback (for API users). ### seekable Controls if seekability is advertised on the file. 0 means non-seekable, -1 means auto (seekable for normal files, non-seekable for named pipes). Many demuxers handle seekable and non-seekable resources differently, overriding this might speed up opening certain files at the cost of losing some features (e.g. accurate seeking). ### ftp FTP (File Transfer Protocol). Read from or write to remote resources using FTP protocol. Following syntax is required. ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg This protocol accepts the following options. ### timeout Set timeout in microseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified. ### ftp-user Set a user to be used for authenticating to the FTP server. This is overridden by the user in the FTP URL. ### ftp-password Set a password to be used for authenticating to the FTP server. This is overridden by the password in the FTP URL, or by **ftp-anonymous-password **if no user is set. ### ftp-anonymous-password Password used when login as anonymous user. Typically an e-mail address should be used. ### ftp-write-seekable Control seekability of connection during encoding. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable. Default value is 0. NOTE: Protocol can be used as output, but it is recommended to not do it, unless special care is taken (tests, customized server configuration etc.). Different FTP servers behave in different way during seek operation. ff* tools may produce incomplete content due to server limitations. ### gopher Gopher protocol. ### gophers Gophers protocol. The Gopher protocol with TLS encapsulation. ### hls Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying "+_proto_" after the hls URI scheme name, where _proto_ is either "file" or "http". hls+ hls+file://path/to/local/resource.m3u8 Using this protocol is discouraged - the hls demuxer should work just as well (if not, please report the issues) and is more complete. To use the hls demuxer instead, simply use the direct URLs to the m3u8 files. ### http HTTP (Hyper Text Transfer Protocol). This protocol accepts the following options: ### seekable Control seekability of connection. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable, if set to -1 it will try to autodetect if it is seekable. Default value is -1. ### chunked_post If set to 1 use chunked Transfer-Encoding for posts, default is 1. ### content_type Set a specific content type for the POST messages or for listen mode. ### http_proxy set HTTP proxy to tunnel through e.g. ### headers Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers. ### multiple_requests Use persistent connections if set to 1, default is 0. ### post_data Set custom HTTP post data. ### referer Set the Referer header. Include 'Referer: URL' header in HTTP request. ### user_agent Override the User-Agent header. If not specified the protocol will use a string describing the libavformat build. ("Lavf/") ### reconnect_at_eof If set then eof is treated like an error and causes reconnection, this is useful for live / endless streams. ### reconnect_streamed If set then even streamed/non seekable streams will be reconnected on errors. ### reconnect_on_network_error Reconnect automatically in case of TCP/TLS errors during connect. ### reconnect_on_http_error A comma separated list of HTTP status codes to reconnect on. The list can include specific status codes (e.g. '503') or the strings '4xx' / '5xx'. ### reconnect_delay_max Sets the maximum delay in seconds after which to give up reconnecting ### mime_type Export the MIME type. ### http_version Exports the HTTP response version number. Usually "1.0" or "1.1". **icy **If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports this, the metadata has to be retrieved by the application by reading the **icy_metadata_headers **and **icy_metadata_packet **options. The default is 1. ### icy_metadata_headers If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers, separated by newline characters. ### icy_metadata_packet If the server supports ICY metadata, and **icy **was set to 1, this contains the last non- empty metadata packet sent by the server. It should be polled in regular intervals by applications interested in mid-stream metadata updates. ### cookies Set the cookies to be sent in future requests. The format of each cookie is the same as the value of a Set-Cookie HTTP response field. Multiple cookies can be delimited by a newline character. ### offset Set initial byte offset. ### end_offset Try to limit the request to bytes preceding this offset. ### method When used as a client option it sets the HTTP method for the request. When used as a server option it sets the HTTP method that is going to be expected from the client(s). If the expected and the received HTTP method do not match the client will be given a Bad Request response. When unset the HTTP method is not checked for now. This will be replaced by autodetection in the future. ### listen If set to 1 enables experimental HTTP server. This can be used to send data when used as an output option, or read data from a client with HTTP POST when used as an input option. If set to 2 enables experimental multi-client HTTP server. This is not yet implemented in ffmpeg.c and thus must not be used as a command line option. # Server side (sending): ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://: # Client side (receiving): ffmpeg -i http://: -c copy somefile.ogg # Client can also be done with wget: wget http://: -O somefile.ogg # Server side (receiving): ffmpeg -listen 1 -i http://: -c copy somefile.ogg # Client side (sending): ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://: # Client can also be done with wget: wget --post-file=somefile.ogg http://: ### send_expect_100 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set to 0 it won't, if set to -1 it will try to send if it is applicable. Default value is -1. ### auth_type Set HTTP authentication type. No option for Digest, since this method requires getting nonce parameters from the server first and can't be used straight away like Basic. **none** Choose the HTTP authentication type automatically. This is the default. **basic** Choose the HTTP basic authentication. Basic authentication sends a Base64-encoded string that contains a user name and password for the client. Base64 is not a form of encryption and should be considered the same as sending the user name and password in clear text (Base64 is a reversible encoding). If a resource needs to be protected, strongly consider using an authentication scheme other than basic authentication. HTTPS/TLS should be used with basic authentication. Without these additional security enhancements, basic authentication should not be used to protect sensitive or valuable information. _HTTP_ _Cookies_ Some HTTP requests will be denied unless cookie values are passed in with the request. The **cookies **option allows these cookies to be specified. At the very least, each cookie must specify a value along with a path and domain. HTTP requests that match both the domain and path will automatically include the cookie value in the HTTP Cookie header field. Multiple cookies can be delimited by a newline. The required syntax to play a stream specifying a cookie is: ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" ### Icecast Icecast protocol (stream to Icecast servers) This protocol accepts the following options: ### ice_genre Set the stream genre. ### ice_name Set the stream name. ### ice_description Set the stream description. ### ice_url Set the stream website URL. ### ice_public Set if the stream should be public. The default is 0 (not public). ### user_agent Override the User-Agent header. If not specified a string of the form "Lavf/" will be used. ### password Set the Icecast mountpoint password. ### content_type Set the stream content type. This must be set if it is different from audio/mpeg. ### legacy_icecast This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT method but the SOURCE method. **tls **Establish a TLS (HTTPS) connection to Icecast. icecast://[[:]@]:/ ### ipfs InterPlanetary File System (IPFS) protocol support. One can access files stored on the IPFS network through so-called gateways. These are http(s) endpoints. This protocol wraps the IPFS native protocols (ipfs:// and ipns://) to be sent to such a gateway. Users can (and should) host their own node which means this protocol will use one's local gateway to access files on the IPFS network. This protocol accepts the following options: ### gateway Defines the gateway to use. When not set, the protocol will first try locating the local gateway by looking at $IPFS_GATEWAY, $IPFS_PATH and "$HOME/.ipfs/", in that order. One can use this protocol in 2 ways. Using IPFS: ffplay ipfs:// Or the IPNS protocol (IPNS is mutable IPFS): ffplay ipns:// ### mmst MMS (Microsoft Media Server) protocol over TCP. ### mmsh MMS (Microsoft Media Server) protocol over HTTP. The required syntax is: mmsh://[:][/][/] ### md5 MD5 output protocol. Computes the MD5 hash of the data to be written, and on close writes this to the designated output or stdout if none is specified. It can be used to test muxers without writing an actual file. Some examples follow. # Write the MD5 hash of the encoded AVI file to the file output.avi.md5. ffmpeg -i input.flv -f avi -y md5:output.avi.md5 # Write the MD5 hash of the encoded AVI file to stdout. ffmpeg -i input.flv -f avi -y md5: Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with the MD5 output protocol. ### pipe UNIX pipe access protocol. Read and write from UNIX pipes. The accepted syntax is: pipe:[] If **fd **isn't specified, _number_ is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If _number_ is not specified, by default the stdout file descriptor will be used for writing, stdin for reading. For example to read from stdin with **ffmpeg**: cat test.wav | ffmpeg -i pipe:0 # ...this is the same as... cat test.wav | ffmpeg -i pipe: For writing to stdout with **ffmpeg**: ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi # ...this is the same as... ffmpeg -i test.wav -f avi pipe: | cat > test.avi This protocol accepts the following options: ### blocksize Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable if data transmission is slow. **fd **Set file descriptor. Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol. ### prompeg Pro-MPEG Code of Practice #3 Release 2 FEC protocol. The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism for MPEG-2 Transport Streams sent over RTP. This protocol must be used in conjunction with the "rtp_mpegts" muxer and the "rtp" protocol. The required syntax is: -f rtp_mpegts -fec prompeg=