Filtering

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

         _______              ______________            ________
        |       |            |              |          |        |
        | 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.

Description

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

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.

Stream specifiers

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

-L

Show license.

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

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

Possible values of arg are:

-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.

-colors

Show recognized color names.

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

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

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

        ffmpeg -sinks pulse,server=192.168.0.4
    

-loglevel [flags+]loglevel | -v [flags+]loglevel

Set logging level and flags used by the library.

The optional flags prefix can consist of the following values:

        ffmpeg -loglevel repeat+level+verbose -i input output

        ffmpeg [...] -loglevel +repeat

-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:

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

-hide_banner

Suppress printing banner.

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

-cpuflags flags (global)

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

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

Possible flags for this option are:

-max_alloc bytes

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

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 fmt (input/output)

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 url (input)

input file url

-y (global)

Overwrite output files without asking.

-n (global)

Do not overwrite output files, and exit immediately if a specified output file already exists.

-stream_loop number (input)

Set number of times input stream shall be looped. Loop 0 means no loop, loop -1 means infinite loop.

-c[:stream_specifier] codec (input/output,per-stream)

-codec[:stream_specifier] codec (input/output,per-stream)

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 duration (input/output)

When used as an input option (before "-i"), limit the duration of data read from the input file.

When used as an output option (before an output url), stop writing the output after its duration reaches duration.

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

-to and -t are mutually exclusive and -t has priority.

-to position (input/output)

Stop writing the output or reading the input at position. position must be a time duration specification, see the Time duration section in the ffmpeg-utils(1) manual.

-to and -t are mutually exclusive and -t has priority.

-fs limit_size (output)

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 position (input/output)

When used as an input option (before "-i"), seeks in this input file to position. Note that in most formats it is not possible to seek exactly, so ffmpeg will seek to the closest seek point before position. When transcoding and -accurate_seek is enabled (the default), this extra segment between the seek point and position will be decoded and discarded. When doing stream copy or when -noaccurate_seek is used, it will be preserved.

When used as an output option (before an output url), decodes but discards input until the timestamps reach position.

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

-sseof position (input)

Like the "-ss" option but relative to the "end of file". That is negative values are earlier in the file, 0 is at EOF.

-itsoffset offset (input)

Set the input time offset.

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

The offset is added to the timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by the time duration specified in offset.

-itsscale scale (input,per-stream)

Rescale input timestamps. scale should be a floating point number.

-timestamp date (output)

Set the recording timestamp in the container.

date must be a date specification, see the Date section in the ffmpeg-utils(1) manual.

-metadata[:metadata_specifier] key=value (output,per-metadata)

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] value (output,per-stream)

Sets the disposition for a stream.

This option overrides the disposition copied from the input stream. It is also possible to delete the disposition by setting it to 0.

The following dispositions are recognized:

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

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

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

-program [title=title:][program_num=program_num:]st=stream[:st=stream...] (output)

Creates a program with the specified title, program_num and adds the specified stream(s) to it.

-target type (output)

Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50"). type may be prefixed with "pal-", "ntsc-" or "film-" to use the corresponding standard. All the format options (bitrate, codecs, buffer sizes) are then set automatically. You can just type:

        ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg
    

Nevertheless you can specify additional options as long as you know they do not conflict with the standard, as in:

        ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
    

The parameters set for each target are as follows.

VCD

        <pal>:
        -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
        -s 352x288 -r 25
        -codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150v -minrate:v 1150k -bufsize:v 327680
        -ar 44100 -ac 2
        -codec:a mp2 -b:a 224k
        
        <ntsc>:
        -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
        -s 352x240 -r 30000/1001
        -codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150v -minrate:v 1150k -bufsize:v 327680
        -ar 44100 -ac 2
        -codec:a mp2 -b:a 224k
        
        <film>:
        -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
        -s 352x240 -r 24000/1001
        -codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150v -minrate:v 1150k -bufsize:v 327680
        -ar 44100 -ac 2
        -codec:a mp2 -b:a 224k
    

SVCD

        <pal>:
        -f svcd -packetsize 2324
        -s 480x576 -pix_fmt yuv420p -r 25
        -codec:v mpeg2video -g 15 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
        -ar 44100
        -codec:a mp2 -b:a 224k
        
        <ntsc>:
        -f svcd -packetsize 2324
        -s 480x480 -pix_fmt yuv420p -r 30000/1001
        -codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
        -ar 44100
        -codec:a mp2 -b:a 224k
        
        <film>:
        -f svcd -packetsize 2324
        -s 480x480 -pix_fmt yuv420p -r 24000/1001
        -codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
        -ar 44100
        -codec:a mp2 -b:a 224k
    

DVD

        <pal>:
        -f dvd -muxrate 10080k -packetsize 2048
        -s 720x576 -pix_fmt yuv420p -r 25
        -codec:v mpeg2video -g 15 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
        -ar 48000
        -codec:a ac3 -b:a 448k
        
        <ntsc>:
        -f dvd -muxrate 10080k -packetsize 2048
        -s 720x480 -pix_fmt yuv420p -r 30000/1001
        -codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
        -ar 48000
        -codec:a ac3 -b:a 448k
        
        <film>:
        -f dvd -muxrate 10080k -packetsize 2048
        -s 720x480 -pix_fmt yuv420p -r 24000/1001
        -codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
        -ar 48000
        -codec:a ac3 -b:a 448k
    

DV

        <pal>:
        -f dv
        -s 720x576 -pix_fmt yuv420p -r 25
        -ar 48000 -ac 2
        
        <ntsc>:
        -f dv
        -s 720x480 -pix_fmt yuv411p -r 30000/1001
        -ar 48000 -ac 2
        
        <film>:
        -f dv
        -s 720x480 -pix_fmt yuv411p -r 24000/1001
        -ar 48000 -ac 2
    

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

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

-dn (input/output)

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 number (output)

Set the number of data frames to output. This is an obsolete alias for "-frames:d", which you should use instead.

-frames[:stream_specifier] framecount (output,per-stream)

Stop writing to the stream after framecount frames.

-q[:stream_specifier] q (output,per-stream)

-qscale[:stream_specifier] q (output,per-stream)

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[:stream_specifier] filtergraph (output,per-stream)

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[:stream_specifier] filename (output,per-stream)

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.

-filter_threads nb_threads (global)

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[:stream_specifier] preset_name (output,per-stream)

Specify the preset for matching stream(s).

-stats (global)

Print encoding progress/statistics. It is on by default, to explicitly disable it you need to specify "-nostats".

-stats_period time (global)

Set period at which encoding progress/statistics are updated. Default is 0.5 seconds.

-progress url (global)

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 (global)

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 filename (output)

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[:stream_specifier] filename (input,per-stream)

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 number (output)

Set the number of video frames to output. This is an obsolete alias for "-frames:v", which you should use instead.

-r[:stream_specifier] fps (input/output,per-stream)

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

As an input option, ignore any timestamps stored in the file and instead generate timestamps assuming constant frame rate fps. This is not the same as the -framerate option used for some input formats like image2 or v4l2 (it used to be the same in older versions of FFmpeg). If in doubt use -framerate instead of the input option -r.

As an output option, duplicate or drop input frames to achieve constant output frame rate fps.

-fpsmax[:stream_specifier] fps (output,per-stream)

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[:stream_specifier] size (input/output,per-stream)

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[:stream_specifier] aspect (output,per-stream)

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.

-vn (input/output)

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 codec (output)

Set the video codec. This is an alias for "-codec:v".

-pass[:stream_specifier] n (output,per-stream)

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[:stream_specifier] prefix (output,per-stream)

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 filtergraph (output)

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[:stream_specifier] format (input/output,per-stream)

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 flags (input/output)

Set SwScaler flags.

-rc_override[:stream_specifier] override (output,per-stream)

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.

-ilme

Force interlacing support in encoder (MPEG-2 and MPEG-4 only). Use this option if your input file is interlaced and you want to keep the interlaced format for minimum losses. The alternative is to deinterlace the input stream by use of a filter such as "yadif" or "bwdif", but deinterlacing introduces losses.

-psnr

Calculate PSNR of compressed frames.

-vstats

Dump video coding statistics to vstats_HHMMSS.log.

-vstats_file file

Dump video coding statistics to file.

-vstats_version file

Specifies which version of the vstats format to use. Default is 2.

version = 1 :

"frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s"

version > 1:

"out= %2d st= %2d frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s"

-top[:stream_specifier] n (output,per-stream)

top=1/bottom=0/auto=-1 field first

-dc precision

Intra_dc_precision.

-vtag fourcc/tag (output)

Force video tag/fourcc. This is an alias for "-tag:v".

-qphist (global)

Show QP histogram

-vbsf bitstream_filter

Deprecated see -bsf

-force_key_frames[:stream_specifier] time[,time...] (output,per-stream)

-force_key_frames[:stream_specifier] expr:expr (output,per-stream)

-force_key_frames[:stream_specifier] source (output,per-stream)

force_key_frames can take arguments of the following form:

        -force_key_frames expr:gte(t,n_forced*5)

        -force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))

-copyinkf[:stream_specifier] (output,per-stream)

When doing stream copy, copy also non-key frames found at the beginning.

-init_hw_device type[=name][:device[,key=value...]]

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:

-init_hw_device type[=name]@source

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 name

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[:stream_specifier] hwaccel (input,per-stream)

Use hardware acceleration to decode the matching stream(s). The allowed values of hwaccel are:

-hwaccel_device[:stream_specifier] hwaccel_device (input,per-stream)

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 methods supported in this build of ffmpeg.

Audio Options

-aframes number (output)

Set the number of audio frames to output. This is an obsolete alias for "-frames:a", which you should use instead.

-ar[:stream_specifier] freq (input/output,per-stream)

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 q (output)

Set the audio quality (codec-specific, VBR). This is an alias for -q:a.

-ac[:stream_specifier] channels (input/output,per-stream)

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 (input/output)

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 codec (input/output)

Set the audio codec. This is an alias for "-codec:a".

-sample_fmt[:stream_specifier] sample_fmt (output,per-stream)

Set the audio sample format. Use "-sample_fmts" to get a list of supported sample formats.

-af filtergraph (output)

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 fourcc/tag (output)

Force audio tag/fourcc. This is an alias for "-tag:a".

-absf bitstream_filter

Deprecated, see -bsf

-guess_layout_max channels (input,per-stream)

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 codec (input/output)

Set the subtitle codec. This is an alias for "-codec:s".

-sn (input/output)

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 bitstream_filter

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 size

Set the size of the canvas used to render subtitles.

Advanced options

-map [-]input_file_id[:stream_specifier][?][,sync_file_id[:stream_specifier]] | [linklabel] (output)

Designate one or more input streams as a source for the output file. Each input stream is identified by the input file index input_file_id and the input stream index input_stream_id within the input file. Both indices start at 0. If specified, sync_file_id:stream_specifier sets which input stream is used as a presentation sync reference.

The first "-map" option on the command line specifies the source for output stream 0, the second "-map" option specifies the source for output stream 1, etc.

A "-" character before the stream identifier creates a "negative" mapping. It disables matching streams from already created mappings.

A trailing "?" after the stream index will allow the map to be optional: if the map matches no streams the map will be ignored instead of failing. Note the map will still fail if an invalid input file index is used; such as if the map refers to a non-existent input.

An alternative [linklabel] form will map outputs from complex filter graphs (see the -filter_complex option) to the output file. linklabel must correspond to a defined output link label in the graph.

For example, to map ALL streams from the first input file to output

        ffmpeg -i INPUT -map 0 output
    

For example, if you have two audio streams in the first input file, these streams are identified by "0:0" and "0:1". You can use "-map" to select which streams to place in an output file. For example:

        ffmpeg -i INPUT -map 0:1 out.wav
    

will map the input stream in INPUT identified by "0:1" to the (single) output stream in out.wav.

For example, to select the stream with index 2 from input file a.mov (specified by the identifier "0:2"), and stream with index 6 from input b.mov (specified by the identifier "1:6"), and copy them to the output file out.mov:

        ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov
    

To select all video and the third audio stream from an input file:

        ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT
    

To map all the streams except the second audio, use negative mappings

        ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
    

To map the video and audio streams from the first input, and using the trailing "?", ignore the audio mapping if no audio streams exist in the first input:

        ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT
    

To pick the English audio stream:

        ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
    

Note that using this option disables the default mappings for this output file.

-ignore_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]

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[:metadata_spec_out] infile[:metadata_spec_in] (output,per-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:

        ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3

        ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv

-map_chapters input_file_index (output)

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 (global)

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 (global)

Show benchmarking information during the encode. Shows real, system and user time used in various steps (audio/video encode/decode).

-timelimit duration (global)

Exit after ffmpeg has been running for duration seconds in CPU user time.

-dump (global)

Dump each input packet to stderr.

-hex (global)

When dumping packets, also dump the payload.

-re (input)

Read input at native frame rate. Mainly used to simulate a grab device, or live input stream (e.g. when reading from a file). Should not be used with actual grab devices or live input streams (where it can cause packet loss). By default ffmpeg attempts to read the input(s) as fast as possible. This option will slow down the reading of the input(s) to the native frame rate of the input(s). It is useful for real-time output (e.g. live streaming).

-vsync parameter

Video sync method. For compatibility reasons old values can be specified as numbers. Newly added values will have to be specified as strings always.

-frame_drop_threshold parameter

Frame drop threshold, which specifies how much behind video frames can be before they are dropped. In frame rate units, so 1.0 is one frame. The default is -1.1. One possible usecase is to avoid framedrops in case of noisy timestamps or to increase frame drop precision in case of exact timestamps.

-async samples_per_second

Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps, the parameter is the maximum samples per second by which the audio is changed. -async 1 is a special case where only the start of the audio stream is corrected without any later correction.

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

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

-adrift_threshold time

Set the minimum difference between timestamps and audio data (in seconds) to trigger adding/dropping samples to make it match the timestamps. This option effectively is a threshold to select between hard (add/drop) and soft (squeeze/stretch) compensation. "-async" must be set to a positive value.

-apad parameters (output,per-stream)

Pad the output audio stream(s). This is the same as applying "-af apad". Argument is a string of filter parameters composed the same as with the "apad" filter. "-shortest" must be set for this output for the option to take effect.

-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 mode

Specify how to set the encoder timebase when stream copying. mode is an integer numeric value, and can assume one of the following values:

-enc_time_base[:stream_specifier] timebase (output,per-stream)

Set the encoder timebase. timebase is a floating point number, and can assume one of the following values:

-bitexact (input/output)

Enable bitexact mode for (de)muxer and (de/en)coder

-shortest (output)

Finish encoding when the shortest input stream ends.

-dts_delta_threshold

Timestamp discontinuity delta threshold.

-dts_error_threshold seconds

Timestamp error delta threshold. This threshold use to discard crazy/damaged timestamps and the default is 30 hours which is arbitrarily picked and quite conservative.

-muxdelay seconds (output)

Set the maximum demux-decode delay.

-muxpreload seconds (output)

Set the initial demux-decode delay.

-streamid output-stream-index:new-value (output)

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[:stream_specifier] bitstream_filters (output,per-stream)

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[:stream_specifier] codec_tag (input/output,per-stream)

Force a tag/fourcc for matching streams.

-timecode hh:mm:ssSEPff

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 filtergraph (global)

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 nb_threads (global)

Defines how many threads are used to process a filter_complex graph. Similar to filter_threads but used for "-filter_complex" graphs only. The default is the number of available CPUs.

-lavfi filtergraph (global)

Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or outputs. Equivalent to -filter_complex.

-filter_complex_script filename (global)

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 (input)

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 (input)

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 size (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 do this if multiple inputs are specified.

-sdp_file file (global)

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 (input)

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.

-abort_on flags (global)

Stop and abort on various conditions. The following flags are available:

-max_error_rate (global)

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 (global)

Stop and exit on error

-max_muxing_queue_size packets (output,per-stream)

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 bytes (output,per-stream)

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 (global)

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".

Preset files

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.

Video and Audio grabbing

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

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

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

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

X11 grabbing

        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

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
      

Quoting and escaping

• ' 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

        [(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

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

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

or

        [-]<S>+[.<m>...][s|ms|us]

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

In both expressions, the optional - indicates negative duration.

Examples

The following examples are all valid time duration:

55

55 seconds

0.2

0.2 seconds

200ms

200 milliseconds, that's 0.2s

200000us

200000 microseconds, that's 0.2s

12:03:45

12 hours, 03 minutes and 45 seconds

23.189

23.189 seconds

Video size

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

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

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

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

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

FL

front left

FR

front right

FC

front center

LFE

low frequency

BL

back left

BR

back right

FLC

front left-of-center

FRC

front right-of-center

BC

back center

SL

side left

SR

side right

TC

top center

TFL

top front left

TFC

top front center

TFR

top front right

TBL

top back left

TBC

top back center

TBR

top back right

DL

downmix left

DR

downmix right

WL

wide left

WR

wide right

SDL

surround direct left

SDR

surround direct right

LFE2

low frequency 2

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

mono

FC

stereo

FL+FR

2.1

FL+FR+LFE

3.0

FL+FR+FC

3.0(back)

FL+FR+BC

4.0

FL+FR+FC+BC

quad

FL+FR+BL+BR

quad(side)

FL+FR+SL+SR

3.1

FL+FR+FC+LFE

5.0

FL+FR+FC+BL+BR

5.0(side)

FL+FR+FC+SL+SR

4.1

FL+FR+FC+LFE+BC

5.1

FL+FR+FC+LFE+BL+BR

5.1(side)

FL+FR+FC+LFE+SL+SR

6.0

FL+FR+FC+BC+SL+SR

6.0(front)

FL+FR+FLC+FRC+SL+SR

hexagonal

FL+FR+FC+BL+BR+BC

6.1

FL+FR+FC+LFE+BC+SL+SR

6.1

FL+FR+FC+LFE+BL+BR+BC

6.1(front)

FL+FR+LFE+FLC+FRC+SL+SR

7.0

FL+FR+FC+BL+BR+SL+SR

7.0(front)

FL+FR+FC+FLC+FRC+SL+SR

7.1

FL+FR+FC+LFE+BL+BR+SL+SR

7.1(wide)

FL+FR+FC+LFE+BL+BR+FLC+FRC

7.1(wide-side)

FL+FR+FC+LFE+FLC+FRC+SL+SR

octagonal

FL+FR+FC+BL+BR+BC+SL+SR

hexadecagonal

FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR

downmix

DL+DR

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

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

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

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

av1

Options

operating_point

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

rawvideo

This decoder decodes rawvideo streams.

Options

top top_field_first

Specify the assumed field type of the input video.

libdav1d

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).

tilethreads

Set amount of tile threads to use during decoding. 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.

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

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

libuavs3d

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).

ac3

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

AC-3 Decoder Options

-drc_scale value

Dynamic Range Scale Factor. The factor to apply to dynamic range values from the AC-3 stream. This factor is applied exponentially. The default value is 1. There are 3 notable scale factor ranges:

flac

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

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 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 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 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 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 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 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.

libaribb24

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

libaribb24 Decoder Options

-aribb24-base-path 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 boolean

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

Enabled by default.

dvbsub

compute_clut

dvb_substream

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

dvdsub

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

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.

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.

aac

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:

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:

ac3 and ac3_fixed

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

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

AC-3 Metadata

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

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

*<<http://www.atsc.org/cms/standards/a_52-2010.pdf>>

*<<http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf>>

*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf>>

*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf>>

Metadata Control Options

-per_frame_metadata boolean

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

Downmix Levels

-center_mixlev level

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

-surround_mixlev level

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

Audio Production Information

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

-mixing_level number

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

-room_type type

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

Other Metadata Options

-copyright boolean

Copyright Indicator. Specifies whether a copyright exists for this audio.

-dialnorm value

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

-dsur_mode mode

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

-original boolean

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

Extended Bitstream Information

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

Extended Bitstream Information - Part 1

-dmix_mode mode

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

-ltrt_cmixlev level

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

-ltrt_surmixlev level

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

-loro_cmixlev level

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

-loro_surmixlev level

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

Extended Bitstream Information - Part 2

-dsurex_mode mode

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

-dheadphone_mode mode

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

-ad_conv_type type

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

Other AC-3 Encoding Options

-stereo_rematrixing boolean

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

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 boolean

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

-cpl_start_band number

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

flac

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

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

ch_mode

Channel mode

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

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

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

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

This encoder has support for the AAC-HE profiles.

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

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

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

Options

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

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:

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:

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:

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

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.

libopencore-amrnb

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.

dtx

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

libopus

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:

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:

cutoff (N.A.)

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

mapping_family (mapping_family)

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

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

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

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

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

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

See also libmp3lame.

Options

The following options are supported by the libshine wrapper. The shineenc-equivalent of the options are listed in parentheses.

b (-b)

Set bitrate expressed in bits/s for CBR. shineenc -b option is expressed in kilobits/s.

libtwolame

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:

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

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.

dtx

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

libvorbis

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

Options

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

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

b (-b)

Set bitrate expressed in bits/s for ABR. oggenc -b is expressed in kilobits/s.

q (-q)

Set constant quality setting for VBR. The value should be a float number in the range of -1.0 to 10.0. The higher the value, the better the quality. The default value is 3.0.

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

cutoff (--advanced-encode-option lowpass_frequency=N)

Set cutoff bandwidth in Hz, a value of 0 disables cutoff. oggenc's related option is expressed in kHz. The default value is 0 (cutoff disabled).

minrate (-m)

Set minimum bitrate expressed in bits/s. oggenc -m is expressed in kilobits/s.

maxrate (-M)

Set maximum bitrate expressed in bits/s. oggenc -M is expressed in kilobits/s. This only has effect on ABR mode.

iblock (--advanced-encode-option impulse_noisetune=N)

Set noise floor bias for impulse blocks. The value is a float number from -15.0 to 0.0. A negative bias instructs the encoder to pay special attention to the crispness of transients in the encoded audio. The tradeoff for better transient response is a higher bitrate.

mjpeg

Options

huffman

Set the huffman encoding strategy. Possible values:

wavpack

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:

optimize_mono

Set whether to enable optimization for mono. This option is only effective for non-mono streams. Available values:

GIF

Options

gifflags integer

Sets the flags used for GIF encoding.

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

Options

format integer

Specifies the Hap format to encode.

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.

jpeg2000

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

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:

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

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

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:

tune (tune)

Set the distortion metric the encoder is tuned with. Default is "psnr".

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:

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