FFM, FFM2 formats¶

FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of video and audio streams and encoding options, and can store a moving time segment of an infinite movie or a whole movie.

FFM is version specific, and there is limited compatibility of FFM files generated by one version of ffmpeg/ffserver and another version of ffmpeg/ffserver. It may work but it is not guaranteed to work.

FFM2 is extensible while maintaining compatibility and should work between differing versions of tools. FFM2 is the default.

Status stream¶

ffserver supports an HTTP interface which exposes the current status of the server.

Simply point your browser to the address of the special status stream specified in the configuration file.

For example if you have:

        <Stream status.html>
        Format status
        
        # Only allow local people to get the status
        ACL allow localhost
        ACL allow 192.168.0.0 192.168.255.255
        </Stream>

then the server will post a page with the status information when the special stream status.html is requested.

How do I make it work?¶

As a simple test, just run the following two command lines where INPUTFILE is some file which you can decode with ffmpeg:

        ffserver -f doc/ffserver.conf &
        ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm

At this point you should be able to go to your Windows machine and fire up Windows Media Player (WMP). Go to Open URL and enter

            http://<linuxbox>:8090/test.asf

You should (after a short delay) see video and hear audio.

WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to transfer the entire file before starting to play. The same is true of AVI files.

You should edit the ffserver.conf file to suit your needs (in terms of frame rates etc). Then install ffserver and ffmpeg, write a script to start them up, and off you go.

What else can it do?¶

You can replay video from .ffm files that was recorded earlier. However, there are a number of caveats, including the fact that the ffserver parameters must match the original parameters used to record the file. If they do not, then ffserver deletes the file before recording into it. (Now that I write this, it seems broken).

You can fiddle with many of the codec choices and encoding parameters, and there are a bunch more parameters that you cannot control. Post a message to the mailing list if there are some 'must have' parameters. Look in ffserver.conf for a list of the currently available controls.

It will automatically generate the ASX or RAM files that are often used in browsers. These files are actually redirections to the underlying ASF or RM file. The reason for this is that the browser often fetches the entire file before starting up the external viewer. The redirection files are very small and can be transferred quickly. [The stream itself is often 'infinite' and thus the browser tries to download it and never finishes.]

Tips¶

* When you connect to a live stream, most players (WMP, RA, etc) want to buffer a certain number of seconds of material so that they can display the signal continuously. However, ffserver (by default) starts sending data in realtime. This means that there is a pause of a few seconds while the buffering is being done by the player. The good news is that this can be cured by adding a '?buffer=5' to the end of the URL. This means that the stream should start 5 seconds in the past -- and so the first 5 seconds of the stream are sent as fast as the network will allow. It will then slow down to real time. This noticeably improves the startup experience.

You can also add a 'Preroll 15' statement into the ffserver.conf that will add the 15 second prebuffering on all requests that do not otherwise specify a time. In addition, ffserver will skip frames until a key_frame is found. This further reduces the startup delay by not transferring data that will be discarded.

Why does the ?buffer / Preroll stop working after a time?¶

It turns out that (on my machine at least) the number of frames successfully grabbed is marginally less than the number that ought to be grabbed. This means that the timestamp in the encoded data stream gets behind realtime. This means that if you say 'Preroll 10', then when the stream gets 10 or more seconds behind, there is no Preroll left.

Fixing this requires a change in the internals of how timestamps are handled.

Does the "?date=" stuff work.¶

Yes (subject to the limitation outlined above). Also note that whenever you start ffserver, it deletes the ffm file (if any parameters have changed), thus wiping out what you had recorded before.

The format of the "?date=xxxxxx" is fairly flexible. You should use one of the following formats (the 'T' is literal):

        * YYYY-MM-DDTHH:MM:SS     (localtime)
        * YYYY-MM-DDTHH:MM:SSZ    (UTC)

You can omit the YYYY-MM-DD, and then it refers to the current day. However note that ?date=16:00:00 refers to 16:00 on the current day -- this may be in the future and so is unlikely to be useful.

You use this by adding the ?date= to the end of the URL for the stream. For example: http://localhost:8080/test.asf?date=2002-07-26T23:05:00.

Stream specifiers¶

Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers are used to precisely specify which stream(s) a given option belongs to.

A stream specifier is a string generally appended to the option name and separated from it by a colon. E.g. "-codec:a:1 ac3" contains the "a:1" stream specifier, which matches the second audio stream. Therefore, it would select the ac3 codec for the second audio stream.

A stream specifier can match several streams, so that the option is applied to all of them. E.g. the stream specifier in "-b:a 128k" matches all audio streams.

An empty stream specifier matches all streams. For example, "-codec copy" or "-codec: copy" would copy all the streams without reencoding.

Possible forms of stream specifiers are:

stream_index

Matches the stream with this index. E.g. "-threads:1 4" would set the thread count for the second stream to 4.

stream_type[:stream_index]

stream_type is one of following: 'v' or 'V' for video, 'a' for audio, 's' for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video streams, 'V' only matches video streams which are not attached pictures, video thumbnails or cover arts. If stream_index is given, then it matches stream number stream_index of this type. Otherwise, it matches all streams of this type.

p:program_id[:stream_index]

If stream_index is given, then it matches the stream with number stream_index in the program with the id program_id. Otherwise, it matches all streams in the program.

#stream_id or i:stream_id

Match the stream by stream id (e.g. PID in MPEG-TS container).

m:key[:value]

Matches streams with the metadata tag key having the specified value. If value is not given, matches streams that contain the given tag with any value.

u

Matches streams with usable configuration, the codec must be defined and the essential information such as video dimension or audio sample rate must be present.

Note that in ffmpeg, matching by metadata will only work properly for input files.

Generic options¶

These options are shared amongst the ff* tools.

-L

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.

-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 [repeat+]loglevel | -v [repeat+]loglevel

Set the logging level used by the library. Adding "repeat+" indicates that repeated log output should not be compressed to the first line and the "Last message repeated n times" line will be omitted. "repeat" can also be used alone. If "repeat" is used alone, and with no prior loglevel set, the default loglevel will be used. If multiple loglevel parameters are given, using 'repeat' will not change the loglevel. loglevel is a string or a number containing one of the following values:

-report

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

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

The following options are recognized:

        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:

-opencl_bench

This option is used to benchmark all available OpenCL devices and print the results. This option is only available when FFmpeg has been compiled with "--enable-opencl".

When FFmpeg is configured with "--enable-opencl", the options for the global OpenCL context are set via -opencl_options. See the "OpenCL Options" section in the ffmpeg-utils manual for the complete list of supported options. Amongst others, these options include the ability to select a specific platform and device to run the OpenCL code on. By default, FFmpeg will run on the first device of the first platform. While the options for the global OpenCL context provide flexibility to the user in selecting the OpenCL device of their choice, most users would probably want to select the fastest OpenCL device for their system.

This option assists the selection of the most efficient configuration by identifying the appropriate device for the user's system. The built-in benchmark is run on all the OpenCL devices and the performance is measured for each device. The devices in the results list are sorted based on their performance with the fastest device listed first. The user can subsequently invoke ffmpeg using the device deemed most appropriate via -opencl_options to obtain the best performance for the OpenCL accelerated code.

Typical usage to use the fastest OpenCL device involve the following steps.

Run the command:

        ffmpeg -opencl_bench
    

Note down the platform ID (pidx) and device ID (didx) of the first i.e. fastest device in the list. Select the platform and device using the command:

        ffmpeg -opencl_options platform_idx=<pidx>:device_idx=<didx> ...
    

-opencl_options options (global)

Set OpenCL environment options. This option is only available when FFmpeg has been compiled with "--enable-opencl".

options must be a list of key=value option pairs separated by ':'. See the ``OpenCL Options'' section in the ffmpeg-utils manual for the list of supported options.

AVOptions¶

These options are provided directly by the libavformat, libavdevice and libavcodec libraries. To see the list of available AVOptions, use the -help option. They are separated into two categories:

generic

These options can be set for any container, codec or device. Generic options are listed under AVFormatContext options for containers/devices and under AVCodecContext options for codecs.

private

These options are specific to the given container, device or codec. Private options are listed under their corresponding containers/devices/codecs.

For example to write an ID3v2.3 header instead of a default ID3v2.4 to an MP3 file, use the id3v2_version private option of the MP3 muxer:

        ffmpeg -i input.flac -id3v2_version 3 out.mp3

All codec AVOptions are per-stream, and thus a stream specifier should be attached to them.

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 configfile

Read configuration file configfile. If not specified it will read by default from /etc/ffserver.conf.

-n

Enable no-launch mode. This option disables all the "Launch" directives within the various "<Feed>" sections. Since ffserver will not launch any ffmpeg instances, you will have to launch them manually.

-d

Enable debug mode. This option increases log verbosity, and directs log messages to stdout. When specified, the CustomLog option is ignored.

ACL syntax¶

An ACL (Access Control List) specifies the address which are allowed to access a given stream, or to write a given feed.

It accepts the following forms

• Allow/deny access to address.

          ACL ALLOW <address>
          ACL DENY <address>
      

• Allow/deny access to ranges of addresses from first_address to last_address.

          ACL ALLOW <first_address> <last_address>
          ACL DENY <first_address> <last_address>
      

You can repeat the ACL allow/deny as often as you like. It is on a per stream basis. The first match defines the action. If there are no matches, then the default is the inverse of the last ACL statement.

Thus 'ACL allow localhost' only allows access from localhost. 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and allow everybody else.

Global options¶

HTTPPort port_number

Port port_number

RTSPPort port_number

HTTPPort sets the HTTP server listening TCP port number, RTSPPort sets the RTSP server listening TCP port number.

Port is the equivalent of HTTPPort and is deprecated.

You must select a different port from your standard HTTP web server if it is running on the same computer.

If not specified, no corresponding server will be created.

HTTPBindAddress ip_address

BindAddress ip_address

RTSPBindAddress ip_address

Set address on which the HTTP/RTSP server is bound. Only useful if you have several network interfaces.

BindAddress is the equivalent of HTTPBindAddress and is deprecated.

MaxHTTPConnections n

Set number of simultaneous HTTP connections that can be handled. It has to be defined before the MaxClients parameter, since it defines the MaxClients maximum limit.

Default value is 2000.

MaxClients n

Set number of simultaneous requests that can be handled. Since ffserver is very fast, it is more likely that you will want to leave this high and use MaxBandwidth.

Default value is 5.

MaxBandwidth kbps

Set the maximum amount of kbit/sec that you are prepared to consume when streaming to clients.

Default value is 1000.

CustomLog filename

Set access log file (uses standard Apache log file format). '-' is the standard output.

If not specified ffserver will produce no log.

In case the commandline option -d is specified this option is ignored, and the log is written to standard output.

NoDaemon

Set no-daemon mode. This option is currently ignored since now ffserver will always work in no-daemon mode, and is deprecated.

UseDefaults

NoDefaults

Control whether default codec options are used for the all streams or not. Each stream may overwrite this setting for its own. Default is UseDefaults. The last occurrence overrides the previous if multiple definitions exist.

Feed section¶

A Feed section defines a feed provided to ffserver.

Each live feed contains one video and/or audio sequence coming from an ffmpeg encoder or another ffserver. This sequence may be encoded simultaneously with several codecs at several resolutions.

A feed instance specification is introduced by a line in the form:

        <Feed FEED_FILENAME>

where FEED_FILENAME specifies the unique name of the FFM stream.

The following options are recognized within a Feed section.

File filename

ReadOnlyFile filename

Set the path where the feed file is stored on disk.

If not specified, the /tmp/FEED.ffm is assumed, where FEED is the feed name.

If ReadOnlyFile is used the file is marked as read-only and it will not be deleted or updated.

Truncate

Truncate the feed file, rather than appending to it. By default ffserver will append data to the file, until the maximum file size value is reached (see FileMaxSize option).

FileMaxSize size

Set maximum size of the feed file in bytes. 0 means unlimited. The postfixes "K" (2^10), "M" (2^20), and "G" (2^30) are recognized.

Default value is 5M.

Launch args

Launch an ffmpeg command when creating ffserver.

args must be a sequence of arguments to be provided to an ffmpeg instance. The first provided argument is ignored, and it is replaced by a path with the same dirname of the ffserver instance, followed by the remaining argument and terminated with a path corresponding to the feed.

When the launched process exits, ffserver will launch another program instance.

In case you need a more complex ffmpeg configuration, e.g. if you need to generate multiple FFM feeds with a single ffmpeg instance, you should launch ffmpeg by hand.

This option is ignored in case the commandline option -n is specified.

ACL spec

Specify the list of IP address which are allowed or denied to write the feed. Multiple ACL options can be specified.

Stream section¶

A Stream section defines a stream provided by ffserver, and identified by a single name.

The stream is sent when answering a request containing the stream name.

A stream section must be introduced by the line:

        <Stream STREAM_NAME>

where STREAM_NAME specifies the unique name of the stream.

The following options are recognized within a Stream section.

Encoding options are marked with the encoding tag, and they are used to set the encoding parameters, and are mapped to libavcodec encoding options. Not all encoding options are supported, in particular it is not possible to set encoder private options. In order to override the encoding options specified by ffserver, you can use the ffmpeg override_ffserver commandline option.

Only one of the Feed and File options should be set.

Feed feed_name

Set the input feed. feed_name must correspond to an existing feed defined in a "Feed" section.

When this option is set, encoding options are used to setup the encoding operated by the remote ffmpeg process.

File filename

Set the filename of the pre-recorded input file to stream.

When this option is set, encoding options are ignored and the input file content is re-streamed as is.

Format format_name

Set the format of the output stream.

Must be the name of a format recognized by FFmpeg. If set to status, it is treated as a status stream.

InputFormat format_name

Set input format. If not specified, it is automatically guessed.

Preroll n

Set this to the number of seconds backwards in time to start. Note that most players will buffer 5-10 seconds of video, and also you need to allow for a keyframe to appear in the data stream.

Default value is 0.

StartSendOnKey

Do not send stream until it gets the first key frame. By default ffserver will send data immediately.

MaxTime n

Set the number of seconds to run. This value set the maximum duration of the stream a client will be able to receive.

A value of 0 means that no limit is set on the stream duration.

ACL spec

Set ACL for the stream.

DynamicACL spec

RTSPOption option

MulticastAddress address

MulticastPort port

MulticastTTL integer

NoLoop

FaviconURL url

Set favicon (favourite icon) for the server status page. It is ignored for regular streams.

Author value

Comment value

Copyright value

Title value

Set metadata corresponding to the option. All these options are deprecated in favor of Metadata.

Metadata key value

Set metadata value on the output stream.

UseDefaults

NoDefaults

Control whether default codec options are used for the stream or not. Default is UseDefaults unless disabled globally.

NoAudio

NoVideo

Suppress audio/video.

AudioCodec codec_name (encoding,audio)

Set audio codec.

AudioBitRate rate (encoding,audio)

Set bitrate for the audio stream in kbits per second.

AudioChannels n (encoding,audio)

Set number of audio channels.

AudioSampleRate n (encoding,audio)

Set sampling frequency for audio. When using low bitrates, you should lower this frequency to 22050 or 11025. The supported frequencies depend on the selected audio codec.

AVOptionAudio [codec:]option value (encoding,audio)

Set generic or private option for audio stream. Private option must be prefixed with codec name or codec must be defined before.

AVPresetAudio preset (encoding,audio)

Set preset for audio stream.

VideoCodec codec_name (encoding,video)

Set video codec.

VideoBitRate n (encoding,video)

Set bitrate for the video stream in kbits per second.

VideoBitRateRange range (encoding,video)

Set video bitrate range.

A range must be specified in the form minrate-maxrate, and specifies the minrate and maxrate encoding options expressed in kbits per second.

VideoBitRateRangeTolerance n (encoding,video)

Set video bitrate tolerance in kbits per second.

PixelFormat pixel_format (encoding,video)

Set video pixel format.

Debug integer (encoding,video)

Set video debug encoding option.

Strict integer (encoding,video)

Set video strict encoding option.

VideoBufferSize n (encoding,video)

Set ratecontrol buffer size, expressed in KB.

VideoFrameRate n (encoding,video)

Set number of video frames per second.

VideoSize (encoding,video)

Set size of the video frame, must be an abbreviation or in the form WxH. See the Video size section in the ffmpeg-utils(1) manual.

Default value is "160x128".

VideoIntraOnly (encoding,video)

Transmit only intra frames (useful for low bitrates, but kills frame rate).

VideoGopSize n (encoding,video)

If non-intra only, an intra frame is transmitted every VideoGopSize frames. Video synchronization can only begin at an intra frame.

VideoTag tag (encoding,video)

Set video tag.

VideoHighQuality (encoding,video)

Video4MotionVector (encoding,video)

BitExact (encoding,video)

Set bitexact encoding flag.

IdctSimple (encoding,video)

Set simple IDCT algorithm.

Qscale n (encoding,video)

Enable constant quality encoding, and set video qscale (quantization scale) value, expressed in n QP units.

VideoQMin n (encoding,video)

VideoQMax n (encoding,video)

Set video qmin/qmax.

VideoQDiff integer (encoding,video)

Set video qdiff encoding option.

LumiMask float (encoding,video)

DarkMask float (encoding,video)

Set lumi_mask/dark_mask encoding options.

AVOptionVideo [codec:]option value (encoding,video)

Set generic or private option for video stream. Private option must be prefixed with codec name or codec must be defined before.

AVPresetVideo preset (encoding,video)

Set preset for video stream.

preset must be the path of a preset file.

Server status stream

A server status stream is a special stream which is used to show statistics about the ffserver operations.

It must be specified setting the option Format to status.

Redirect section¶

A redirect section specifies where to redirect the requested URL to another page.

A redirect section must be introduced by the line:

        <Redirect NAME>

where NAME is the name of the page which should be redirected.

It only accepts the option URL, which specify the redirection URL.

Quoting and escaping¶

FFmpeg adopts the following quoting and escaping mechanism, unless explicitly specified. The following rules are applied:

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

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

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

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

Examples

• Escape the string "Crime d'Amour" containing the "'" special character:

          Crime d\'Amour
      

• The string above contains a quote, so the "'" needs to be escaped when quoting it:

          'Crime d'\''Amour'
      

• Include leading or trailing whitespaces using quoting:

          '  this string starts and ends with whitespaces  '
      

• Escaping and quoting can be mixed together:

          ' The string '\'string\'' is a string '
      

• To include a literal \ you can use either escaping or quoting:

          'c:\foo' can be written as c:\\foo
      

Date¶

The accepted syntax is:

        [(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
        now

If the value is "now" it takes the current time.

Time is local time unless Z is appended, in which case it is interpreted as UTC. If the year-month-day part is not specified it takes the current year-month-day.

Time duration¶

There are two accepted syntaxes for expressing time duration.

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

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

or

        [-]<S>+[.<m>...]

S expresses the number of seconds, with the optional decimal part m.

In both expressions, the optional - indicates negative duration.

Examples

The following examples are all valid time duration:

55

55 seconds

12:03:45

12 hours, 03 minutes and 45 seconds

23.189

23.189 seconds

Video size¶

Specify the size of the sourced video, it may be a string of the form widthxheight, or the name of a size abbreviation.

The following abbreviations are recognized:

ntsc

720x480

pal

720x576

qntsc

352x240

qpal

352x288

sntsc

640x480

spal

768x576

film

352x240

ntsc-film

352x240

sqcif

128x96

qcif

176x144

cif

352x288

4cif

704x576

16cif

1408x1152

qqvga

160x120

qvga

320x240

vga

640x480

svga

800x600

xga

1024x768

uxga

1600x1200

qxga

2048x1536

sxga

1280x1024

qsxga

2560x2048

hsxga

5120x4096

wvga

852x480

wxga

1366x768

wsxga

1600x1024

wuxga

1920x1200

woxga

2560x1600

wqsxga

3200x2048

wquxga

3840x2400

whsxga

6400x4096

whuxga

7680x4800

cga

320x200

ega

640x350

hd480

852x480

hd720

1280x720

hd1080

1920x1080

2k

2048x1080

2kflat

1998x1080

2kscope

2048x858

4k

4096x2160

4kflat

3996x2160

4kscope

4096x1716

nhd

640x360

hqvga

240x160

wqvga

400x240

fwqvga

432x240

hvga

480x320

qhd

960x540

2kdci

2048x1080

4kdci

4096x2160

uhd2160

3840x2160

uhd4320

7680x4320

Video rate¶

Specify the frame rate of a video, expressed as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation.

The following abbreviations are recognized:

ntsc

30000/1001

pal

25/1

qntsc

30000/1001

qpal

25/1

sntsc

30000/1001

spal

25/1

film

24/1

ntsc-film

24000/1001

Ratio¶

A ratio can be expressed as an expression, or in the form numerator:denominator.

Note that a ratio with infinite (1/0) or negative value is considered valid, so you should check on the returned value if you want to exclude those values.

The undefined value can be expressed using the "0:0" string.

Color¶

It can be the name of a color as defined below (case insensitive match) or a "[0x|#]RRGGBB[AA]" sequence, possibly followed by @ and a string representing the alpha component.

The alpha component may be a string composed by "0x" followed by an hexadecimal number or a decimal number between 0.0 and 1.0, which represents the opacity value (0x00 or 0.0 means completely transparent, 0xff or 1.0 completely opaque). If the alpha component is not specified then 0xff is assumed.

The string random will result in a random color.

The following names of colors are recognized:

AliceBlue

0xF0F8FF

AntiqueWhite

0xFAEBD7

Aqua

0x00FFFF

Aquamarine

0x7FFFD4

Azure

0xF0FFFF

Beige

0xF5F5DC

Bisque

0xFFE4C4

Black

0x000000

BlanchedAlmond

0xFFEBCD

Blue

0x0000FF

BlueViolet

0x8A2BE2

Brown

0xA52A2A

BurlyWood

0xDEB887

CadetBlue

0x5F9EA0

Chartreuse

0x7FFF00

Chocolate

0xD2691E

Coral

0xFF7F50

CornflowerBlue

0x6495ED

Cornsilk

0xFFF8DC

Crimson

0xDC143C

Cyan

0x00FFFF

DarkBlue

0x00008B

DarkCyan

0x008B8B

DarkGoldenRod

0xB8860B

DarkGray

0xA9A9A9

DarkGreen

0x006400

DarkKhaki

0xBDB76B

DarkMagenta

0x8B008B

DarkOliveGreen

0x556B2F

Darkorange

0xFF8C00

DarkOrchid

0x9932CC

DarkRed

0x8B0000

DarkSalmon

0xE9967A

DarkSeaGreen

0x8FBC8F

DarkSlateBlue

0x483D8B

DarkSlateGray

0x2F4F4F

DarkTurquoise

0x00CED1

DarkViolet

0x9400D3

DeepPink

0xFF1493

DeepSkyBlue

0x00BFFF

DimGray

0x696969

DodgerBlue

0x1E90FF

FireBrick

0xB22222

FloralWhite

0xFFFAF0

ForestGreen

0x228B22

Fuchsia

0xFF00FF

Gainsboro

0xDCDCDC

GhostWhite

0xF8F8FF

Gold

0xFFD700

GoldenRod

0xDAA520

Gray

0x808080

Green

0x008000

GreenYellow

0xADFF2F

HoneyDew

0xF0FFF0

HotPink

0xFF69B4

IndianRed

0xCD5C5C

Indigo

0x4B0082

Ivory

0xFFFFF0

Khaki

0xF0E68C

Lavender

0xE6E6FA

LavenderBlush

0xFFF0F5

LawnGreen

0x7CFC00

LemonChiffon

0xFFFACD

LightBlue

0xADD8E6

LightCoral

0xF08080

LightCyan

0xE0FFFF

LightGoldenRodYellow

0xFAFAD2

LightGreen

0x90EE90

LightGrey

0xD3D3D3

LightPink

0xFFB6C1

LightSalmon

0xFFA07A

LightSeaGreen

0x20B2AA

LightSkyBlue

0x87CEFA

LightSlateGray

0x778899

LightSteelBlue

0xB0C4DE

LightYellow

0xFFFFE0

Lime

0x00FF00

LimeGreen

0x32CD32

Linen

0xFAF0E6

Magenta

0xFF00FF

Maroon

0x800000

MediumAquaMarine

0x66CDAA

MediumBlue

0x0000CD

MediumOrchid

0xBA55D3

MediumPurple

0x9370D8

MediumSeaGreen

0x3CB371

MediumSlateBlue

0x7B68EE

MediumSpringGreen

0x00FA9A

MediumTurquoise

0x48D1CC

MediumVioletRed

0xC71585

MidnightBlue

0x191970

MintCream

0xF5FFFA

MistyRose

0xFFE4E1

Moccasin

0xFFE4B5

NavajoWhite

0xFFDEAD

Navy

0x000080

OldLace

0xFDF5E6

Olive

0x808000

OliveDrab

0x6B8E23

Orange

0xFFA500

OrangeRed

0xFF4500

Orchid

0xDA70D6

PaleGoldenRod

0xEEE8AA

PaleGreen

0x98FB98

PaleTurquoise

0xAFEEEE

PaleVioletRed

0xD87093

PapayaWhip

0xFFEFD5

PeachPuff

0xFFDAB9

Peru

0xCD853F

Pink

0xFFC0CB

Plum

0xDDA0DD

PowderBlue

0xB0E0E6

Purple

0x800080

Red

0xFF0000

RosyBrown

0xBC8F8F

RoyalBlue

0x4169E1

SaddleBrown

0x8B4513

Salmon

0xFA8072

SandyBrown

0xF4A460

SeaGreen

0x2E8B57

SeaShell

0xFFF5EE

Sienna

0xA0522D

Silver

0xC0C0C0

SkyBlue

0x87CEEB

SlateBlue

0x6A5ACD

SlateGray

0x708090

Snow

0xFFFAFA

SpringGreen

0x00FF7F

SteelBlue

0x4682B4

Tan

0xD2B48C

Teal

0x008080

Thistle

0xD8BFD8

Tomato

0xFF6347

Turquoise

0x40E0D0

Violet

0xEE82EE

Wheat

0xF5DEB3

White

0xFFFFFF

WhiteSmoke

0xF5F5F5

Yellow

0xFFFF00

YellowGreen

0x9ACD32

Channel Layout¶

A channel layout specifies the spatial disposition of the channels in a multi-channel audio stream. To specify a channel layout, FFmpeg makes use of a special syntax.

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

FL

front left

FR

front right

FC

front center

LFE

low frequency

BL

back left

BR

back right

FLC

front left-of-center

FRC

front right-of-center

BC

back center

SL

side left

SR

side right

TC

top center

TFL

top front left

TFC

top front center

TFR

top front right

TBL

top back left

TBC

top back center

TBR

top back right

DL

downmix left

DR

downmix right

WL

wide left

WR

wide right

SDL

surround direct left

SDR

surround direct right

LFE2

low frequency 2

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

mono

FC

stereo

FL+FR

2.1

FL+FR+LFE

3.0

FL+FR+FC

3.0(back)

FL+FR+BC

4.0

FL+FR+FC+BC

quad

FL+FR+BL+BR

quad(side)

FL+FR+SL+SR

3.1

FL+FR+FC+LFE

5.0

FL+FR+FC+BL+BR

5.0(side)

FL+FR+FC+SL+SR

4.1

FL+FR+FC+LFE+BC

5.1

FL+FR+FC+LFE+BL+BR

5.1(side)

FL+FR+FC+LFE+SL+SR

6.0

FL+FR+FC+BC+SL+SR

6.0(front)

FL+FR+FLC+FRC+SL+SR

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

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.

hevc¶

HEVC / H.265 decoder.

Note: the skip_loop_filter option has effect only at level "all".

rawvideo¶

Raw video decoder.

This decoder decodes rawvideo streams.

Options

top top_field_first

Specify the assumed field type of the input video.

ac3¶

AC-3 audio decoder.

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

AC-3 Decoder Options

-drc_scale value

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

flac¶

FLAC audio decoder.

This decoder aims to implement the complete FLAC specification from Xiph.

FLAC Decoder options

-use_buggy_lpc

The lavc FLAC encoder used to produce buggy streams with high lpc values (like the default value). This option makes it possible to decode such streams correctly by using lavc's old buggy lpc logic for decoding.

ffwavesynth¶

Internal wave synthesizer.

This decoder generates wave patterns according to predefined sequences. Its use is purely internal and the format of the data it accepts is not publicly documented.

libcelt¶

libcelt decoder wrapper.

libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec. Requires the presence of the libcelt headers and library during configuration. You need to explicitly configure the build with "--enable-libcelt".

libgsm¶

libgsm decoder wrapper.

libgsm allows libavcodec to decode the GSM full rate audio codec. Requires the presence of the libgsm headers and library during configuration. You need to explicitly configure the build with "--enable-libgsm".

This decoder supports both the ordinary GSM and the Microsoft variant.

libilbc¶

libilbc decoder wrapper.

libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC) audio codec. Requires the presence of the libilbc headers and library during configuration. You need to explicitly configure the build with "--enable-libilbc".

Options

The following option is supported by the libilbc wrapper.

enhance

Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled).

libopencore-amrnb¶

libopencore-amrnb decoder wrapper.

libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate Narrowband audio codec. Using it requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with "--enable-libopencore-amrnb".

An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB without this library.

libopencore-amrwb¶

libopencore-amrwb decoder wrapper.

libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate Wideband audio codec. Using it requires the presence of the libopencore-amrwb headers and library during configuration. You need to explicitly configure the build with "--enable-libopencore-amrwb".

An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB without this library.

libopus¶

libopus decoder wrapper.

libopus allows libavcodec to decode the Opus Interactive Audio Codec. Requires the presence of the libopus headers and library during configuration. You need to explicitly configure the build with "--enable-libopus".

An FFmpeg native decoder for Opus exists, so users can decode Opus without this library.

dvbsub¶

Options

compute_clut

dvb_substream

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

dvdsub¶

This codec decodes the bitmap subtitles used in DVDs; the same subtitles can also be found in VobSub file pairs and in some Matroska files.

Options

palette

Specify the global palette used by the bitmaps. When stored in VobSub, the palette is normally specified in the index file; in Matroska, the palette is stored in the codec extra-data in the same format as in VobSub. In DVDs, the palette is stored in the IFO file, and therefore not available when reading from dumped VOB files.

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

ifo_palette

Specify the IFO file from which the global palette is obtained. (experimental)

forced_subs_only

Only decode subtitle entries marked as forced. Some titles have forced and non-forced subtitles in the same track. Setting this flag to 1 will only keep the forced subtitles. Default value is 0.

libzvbi-teletext¶

Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext subtitles. Requires the presence of the libzvbi headers and library during configuration. You need to explicitly configure the build with "--enable-libzvbi".

Options

txt_page

List of teletext page numbers to decode. You may use the special * string to match all pages. Pages that do not match the specified list are dropped. Default value is *.

txt_chop_top

Discards the top teletext line. Default value is 1.

txt_format

Specifies the format of the decoded subtitles. The teletext decoder is capable of decoding the teletext pages to bitmaps or to simple text, you should use "bitmap" for teletext pages, because certain graphics and colors cannot be expressed in simple text. You might use "text" for teletext based subtitles if your application can handle simple text based subtitles. Default value is bitmap.

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 30000 which is 30 seconds.

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¶

Advanced Audio Coding (AAC) encoder.

This encoder is the default AAC encoder, natively implemented into FFmpeg. Its quality is on par or better than libfdk_aac at the default bitrate of 128kbps. This encoder also implements more options, profiles and samplerates than other encoders (with only the AAC-HE profile pending to be implemented) so this encoder has become the default and is the recommended choice.

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¶

AC-3 audio encoders.

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

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

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¶

FLAC (Free Lossless Audio Codec) Encoder

Options

The following options are supported by FFmpeg's flac encoder.

compression_level

Sets the compression level, which chooses defaults for many other options if they are not set explicitly. Valid values are from 0 to 12, 5 is the default.

frame_size

Sets the size of the frames in samples per channel.

lpc_coeff_precision

Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the default.

lpc_type

Sets the first stage LPC algorithm

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¶

Opus encoder.

This is a native FFmpeg encoder for the Opus format. Currently its in development and only implements the CELT part of the codec. Its quality is usually worse and at best is equal to the libopus encoder.

Options

b

Set bit rate in bits/s. If unspecified it uses the number of channels and the layout to make a good guess.

opus_delay

Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly decrease quality.

libfdk_aac¶

libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.

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

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

This encoder is considered to produce output on par or worse at 128kbps to the the native FFmpeg AAC encoder but can often produce better sounding audio at identical or lower bitrates and has support for the AAC-HE profiles.

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

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

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

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.

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¶

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

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

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

Options

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

b (-b)

Set bitrate expressed in bits/s for CBR or ABR. LAME "bitrate" is expressed in kilobits/s.

q (-V)

Set constant quality setting for VBR. This option is valid only using the ffmpeg command-line tool. For library interface users, use global_quality.

compression_level (-q)

Set algorithm quality. Valid arguments are integers in the 0-9 range, with 0 meaning highest quality but slowest, and 9 meaning fastest while producing the worst quality.

cutoff (--lowpass)

Set lowpass cutoff frequency. If unspecified, the encoder dynamically adjusts the cutoff.

reservoir

Enable use of bit reservoir when set to 1. Default value is 1. LAME has this enabled by default, but can be overridden by use --nores option.

joint_stereo (-m j)

Enable the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo. Default value is 1.

abr (--abr)

Enable the encoder to use ABR when set to 1. The lame --abr sets the target bitrate, while this options only tells FFmpeg to use ABR still relies on b to set bitrate.

libopencore-amrnb¶

OpenCORE Adaptive Multi-Rate Narrowband encoder.

Requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with "--enable-libopencore-amrnb --enable-version3".

This is a mono-only encoder. Officially it only supports 8000Hz sample rate, but you can override it by setting strict to unofficial or lower.

Options

b

Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.

dtx

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

libopus¶

libopus Opus Interactive Audio Codec encoder wrapper.

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

Option Mapping

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

b (bitrate)

Set the bit rate in bits/s. FFmpeg's b option is expressed in bits/s, while opusenc's bitrate in kilobits/s.

vbr (vbr, hard-cbr, and cvbr)

Set VBR mode. The FFmpeg vbr option has the following valid arguments, with the opusenc equivalent options in parentheses:

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.

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.

libshine¶

Shine Fixed-Point MP3 encoder wrapper.

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

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

The original project (last updated in early 2007) is at <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¶

TwoLAME MP2 encoder wrapper.

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

Options

The following options are supported by the libtwolame wrapper. The twolame-equivalent options follow the FFmpeg ones and are in parentheses.

b (-b)

Set bitrate expressed in bits/s for CBR. twolame b option is expressed in kilobits/s. Default value is 128k.

q (-V)

Set quality for experimental VBR support. Maximum value range is from -50 to 50, useful range is from -10 to 10. The higher the value, the better the quality. This option is valid only using the ffmpeg command-line tool. For library interface users, use global_quality.

mode (--mode)

Set the mode of the resulting audio. Possible values:

psymodel (--psyc-mode)

Set psychoacoustic model to use in encoding. The argument must be an integer between -1 and 4, inclusive. The higher the value, the better the quality. The default value is 3.

energy_levels (--energy)

Enable energy levels extensions when set to 1. The default value is 0 (disabled).

error_protection (--protect)

Enable CRC error protection when set to 1. The default value is 0 (disabled).

copyright (--copyright)

Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).

original (--original)

Set MPEG audio original flag when set to 1. The default value is 0 (disabled).

libvo-amrwbenc¶

VisualOn Adaptive Multi-Rate Wideband encoder.

Requires the presence of the libvo-amrwbenc headers and library during configuration. You need to explicitly configure the build with "--enable-libvo-amrwbenc --enable-version3".

This is a mono-only encoder. Officially it only supports 16000Hz sample rate, but you can override it by setting strict to unofficial or lower.

Options

b

Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.

dtx

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

libvorbis¶

libvorbis encoder wrapper.

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

Options

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

To get a more accurate and extensive documentation of the libvorbis options, consult the libvorbisenc's and oggenc's documentations. See <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.

libwavpack¶

A wrapper providing WavPack encoding through libwavpack.

Only lossless mode using 32-bit integer samples is supported currently.

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

Note that a libavcodec-native encoder for the WavPack codec exists so users can encode audios with this codec without using this encoder. See wavpackenc.

Options

wavpack command line utility's corresponding options are listed in parentheses, if any.

frame_size (--blocksize)

Default is 32768.

compression_level

Set speed vs. compression tradeoff. Acceptable arguments are listed below:

mjpeg¶

Motion JPEG encoder.

Options

huffman

Set the huffman encoding strategy. Possible values:

wavpack¶

WavPack lossless audio encoder.

This is a libavcodec-native WavPack encoder. There is also an encoder based on libwavpack, but there is virtually no reason to use that encoder.

See also libwavpack.

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)

This option's syntax is consistent with libwavpack's.

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:

Hap¶

Vidvox Hap video encoder.

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¶

The native jpeg 2000 encoder is lossy by default, the "-q:v" option can be used to set the encoding quality. Lossless encoding can be selected with "-pred 1".

Options

format

Can be set to either "j2k" or "jp2" (the default) that makes it possible to store non-rgb pix_fmts.

libkvazaar¶

Kvazaar H.265/HEVC encoder.

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

Options

b

Set target video bitrate in bit/s and enable rate control.

kvazaar-params

Set kvazaar parameters as a list of name=value pairs separated by commas (,). See kvazaar documentation for a list of options.

libopenh264¶

Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.

This encoder requires the presence of the libopenh264 headers and library during configuration. You need to explicitly configure the build with "--enable-libopenh264". The library is detected using pkg-config.

For more information about the library see <http://www.openh264.org>.

Options

The following FFmpeg global options affect the configurations of the libopenh264 encoder.

b

Set the bitrate (as a number of bits per second).

g

Set the GOP size.

maxrate

Set the max bitrate (as a number of bits per second).

flags +global_header

Set global header in the bitstream.

slices

Set the number of slices, used in parallelized encoding. Default value is 0. This is only used when slice_mode is set to fixed.

slice_mode

Set slice mode. Can assume one of the following possible values:

loopfilter

Enable loop filter, if set to 1 (automatically enabled). To disable set a value of 0.

profile

Set profile restrictions. If set to the value of main enable CABAC (set the "SEncParamExt.iEntropyCodingModeFlag" flag to 1).

max_nal_size

Set maximum NAL size in bytes.

allow_skip_frames

Allow skipping frames to hit the target bitrate if set to 1.

libtheora¶

libtheora Theora encoder wrapper.

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

For more information about the libtheora project see <http://www.theora.org/>.

Options

The following global options are mapped to internal libtheora options which affect the quality and the bitrate of the encoded stream.

b

Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In case VBR (Variable Bit Rate) mode is enabled this option is ignored.

flags

Used to enable constant quality mode (VBR) encoding through the qscale flag, and to enable the "pass1" and "pass2" modes.

g

Set the GOP size.

global_quality

Set the global quality as an integer in lambda units.

Only relevant when VBR mode is enabled with "flags +qscale". The value is converted to QP units by dividing it by "FF_QP2LAMBDA", clipped in the [0 - 10] range, and then multiplied by 6.3 to get a value in the native libtheora range [0-63]. A higher value corresponds to a higher quality.

q

Enable VBR mode when set to a non-negative value, and set constant quality value as a double floating point value in QP units.

The value is clipped in the [0-10] range, and then multiplied by 6.3 to get a value in the native libtheora range [0-63].

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

Examples

• Set maximum constant quality (VBR) encoding with ffmpeg:

          ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
      

• Use ffmpeg to convert a CBR 1000 kbps Theora video stream:

          ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
      

libvpx¶

VP8/VP9 format supported through libvpx.

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

Options

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

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

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

b (target-bitrate)

Set bitrate in bits/s. Note that FFmpeg's b option is expressed in bits/s, while vpxenc's target-bitrate is in kilobits/s.

g (kf-max-dist)

keyint_min (kf-min-dist)

qmin (min-q)

qmax (max-q)

bufsize (buf-sz, buf-optimal-sz)

Set ratecontrol buffer size (in bits). Note vpxenc's options are specified in milliseconds, the libvpx wrapper converts this value as follows: "buf-sz = bufsize * 1000 / bitrate", "buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6".

rc_init_occupancy (buf-initial-sz)

Set number of bits which should be loaded into the rc buffer before decoding starts. Note vpxenc's option is specified in milliseconds, the libvpx wrapper converts this value as follows: "rc_init_occupancy * 1000 / bitrate".

undershoot-pct

Set datarate undershoot (min) percentage of the target bitrate.

overshoot-pct

Set datarate overshoot (max) percentage of the target bitrate.

skip_threshold (drop-frame)

qcomp (bias-pct)

maxrate (maxsection-pct)

Set GOP max bitrate in bits/s. Note vpxenc's option is specified as a percentage of the target bitrate, the libvpx wrapper converts this value as follows: "(maxrate * 100 / bitrate)".

minrate (minsection-pct)

Set GOP min bitrate in bits/s. Note vpxenc's option is specified as a percentage of the target bitrate, the libvpx wrapper converts this value as follows: "(minrate * 100 / bitrate)".

minrate, maxrate, b end-usage=cbr

"(minrate == maxrate == bitrate)".

crf (end-usage=cq, cq-level)

tune (tune)

quality, deadline (deadline)

speed, cpu-used (cpu-used)

Set quality/speed ratio modifier. Higher values speed up the encode at the cost of quality.

nr (noise-sensitivity)

static-thresh

Set a change threshold on blocks below which they will be skipped by the encoder.

slices (token-parts)

Note that FFmpeg's slices option gives the total number of partitions, while vpxenc's token-parts is given as "log2(partitions)".

max-intra-rate

Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0 means unlimited.

force_key_frames

"VPX_EFLAG_FORCE_KF"

Alternate reference frame related