Man1 - ffplay-all.1

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

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

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

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

Possible forms of stream specifiers are:

stream_index

Matches the stream with this index. E.g. -threads:1 4 would set the thread count for the second stream to 4. If stream_index is used as an additional stream specifier (see below), then it selects stream number stream_index from the matching streams. Stream numbering is based on the order of the streams as detected by libavformat except when a program ID is also specified. In this case it is based on the ordering of the streams in the program.

stream_type[:additional_stream_specifier]

stream_type is one of following: ’v’ or ’V’ for video, ’a’ for audio, ’s’ for subtitle, ’d’ for data, and ’t’ for attachments. ’v’ matches all video streams, ’V’ only matches video streams which are not attached pictures, video thumbnails or cover arts. If additional_stream_specifier is used, then it matches streams which both have this type and match the additional_stream_specifier. Otherwise, it matches all streams of the specified type.

p:program_id[:additional_stream_specifier]

Matches streams which are in the program with the id program_id. If additional_stream_specifier is used, then it matches streams which both are part of the program and match the additional_stream_specifier.

#stream_id or i:stream_id

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

m:key[:value]

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

(no term)

Matches streams with usable configuration, the codec must be defined and the essential information such as video dimension or audio sample rate must be present. Note that in ffmpeg, matching by metadata will only work properly for input files.

These options are shared amongst the ff* tools.

-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:

long

Print advanced tool options in addition to the basic tool options.

full

Print complete list of options, including shared and private options for encoders, decoders, demuxers, muxers, filters, etc.

decoder=decoder_name

Print detailed information about the decoder named decoder_name. Use the -decoders option to get a list of all decoders.

encoder=encoder_name

Print detailed information about the encoder named encoder_name. Use the -encoders option to get a list of all encoders.

demuxer=demuxer_name

Print detailed information about the demuxer named demuxer_name. Use the -formats option to get a list of all demuxers and muxers.

muxer=muxer_name

Print detailed information about the muxer named muxer_name. Use the -formats option to get a list of all muxers and demuxers.

filter=filter_name

Print detailed information about the filter named filter_name. Use the -filters option to get a list of all filters.

bsf=bitstream_filter_name

Print detailed information about the bitstream filter named bitstream_filter_name. Use the -bsfs option to get a list of all bitstream filters.

protocol=protocol_name

Print detailed information about the protocol named protocol_name. Use the -protocols option to get a list of all protocols.

-version

Show version.

-buildconf

Show the build configuration, one option per line.

-formats

Show available formats (including devices).

-demuxers

Show available demuxers.

-muxers

Show available muxers.

-devices

Show available devices.

-codecs

Show all codecs known to libavcodec. Note that the term ’codec’ is used throughout this documentation as a shortcut for what is more correctly called a media bitstream format.

-decoders

Show available decoders.

-encoders

Show all available encoders.

-bsfs

Show available bitstream filters.

-protocols

Show available protocols.

-filters

Show available libavfilter filters.

-pix_fmts

Show available pixel formats.

-sample_fmts

Show available sample formats.

-layouts

Show channel names and standard channel layouts.

-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:

repeat

Indicates that repeated log output should not be compressed to the first line and the Last message repeated n times line will be omitted.

level

Indicates that log output should add a [level] prefix to each message line. This can be used as an alternative to log coloring, e.g. when dumping the log to file.

Flags can also be used alone by adding a ’’’-’ prefix to set/reset a single flag without affecting other /flags or changing loglevel. When setting both flags and loglevel, a ’’ separator is expected between the last flags value and before loglevel. loglevel is a string or a number containing one of the following values:

quiet, -8

Show nothing at all; be silent.

panic, 0

Only show fatal errors which could lead the process to crash, such as an assertion failure. This is not currently used for anything.

fatal, 8

Only show fatal errors. These are errors after which the process absolutely cannot continue.

error, 16

Show all errors, including ones which can be recovered from.

warning, 24

Show all warnings and errors. Any message related to possibly incorrect or unexpected events will be shown.

info, 32

Show informative messages during processing. This is in addition to warnings and errors. This is the default value.

verbose, 40

Same as info, except more verbose.

debug, 48

Show everything, including debugging information.

trace, 56

For example to enable repeated log output, add the level prefix, and set loglevel to verbose: ffmpeg -loglevel repeat+level+verbose -i input output Another example that enables repeated log output without affecting current state of level prefix flag or loglevel: ffmpeg […] -loglevel +repeat By default the program logs to stderr. If coloring is supported by the terminal, colors are used to mark errors and warnings. Log coloring can be disabled setting the environment variable AV_LOG_FORCE_NOCOLOR, or can be forced setting the environment variable AV_LOG_FORCE_COLOR.

-report

Dump full command line and log output to a file named /=program=/=-/=YYYYMMDD=/-/=HHMMSS=/.log= in the current directory. This file can be useful for bug reports. It also implies -loglevel debug. Setting the environment variable FFREPORT to any value has the same effect. If the value is a ’:’-separated key=value sequence, these options will affect the report; option values must be escaped if they contain special characters or the options delimiter ’:’ (see the ``Quoting and escaping’’ section in the ffmpeg-utils manual). The following options are recognized:

file

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

level

set the log verbosity level using a numerical value (see -loglevel).

For example, to output a report to a file named ffreport.log using a log level of 32 (alias for log level info): FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output Errors in parsing the environment variable are not fatal, and will not appear in the report.

-hide_banner

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

-cpuflags flags (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:

x86

ARM

armv5te

armv6

armv6t2

vfp

vfpv3

neon

setend

AArch64

armv8

vfp

neon

PowerPC

altivec

Specific Processors

pentium2

pentium3

pentium4

k6

k62

athlon

athlonxp

k8

-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.

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

generic

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

private

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

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

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

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

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

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

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

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

-x width

Force displayed width.

-y height

Force displayed height.

-s size

Set frame size (WxH or abbreviation), needed for videos which do not contain a header with the frame size like raw YUV. This option has been deprecated in favor of private options, try -video_size.

-fs

Start in fullscreen mode.

-an

Disable audio.

-vn

Disable video.

-sn

Disable subtitles.

-ss pos

Seek to pos. Note that in most formats it is not possible to seek exactly, so ffplay will seek to the nearest seek point to pos. pos must be a time duration specification, see the Time duration section in the ffmpeg-utils (1) manual.

-t duration

Play duration seconds of audio/video. duration must be a time duration specification, see the Time duration section in the ffmpeg-utils (1) manual.

-bytes

Seek by bytes.

-seek_interval

Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.

-nodisp

Disable graphical display.

-noborder

Borderless window.

-alwaysontop

Window always on top. Available on: X11 with SDL >= 2.0.5, Windows SDL >= 2.0.6.

-volume

Set the startup volume. 0 means silence, 100 means no volume reduction or amplification. Negative values are treated as 0, values above 100 are treated as 100.

-f fmt

Force format.

-window_title title

Set window title (default is the input filename).

-left title

Set the x position for the left of the window (default is a centered window).

-top title

Set the y position for the top of the window (default is a centered window).

-loop number

Loops movie playback <number> times. 0 means forever.

-showmode mode

Set the show mode to use. Available values for mode are:

0, video

show video

1, waves

show audio waves

2, rdft

show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)

Default value is video, if video is not present or cannot be played rdft is automatically selected. You can interactively cycle through the available show modes by pressing the key w.

-vf filtergraph

Create the filtergraph specified by filtergraph and use it to filter the video stream. filtergraph is a description of the filtergraph to apply to the stream, and must have a single video input and a single video output. 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. You can specify this parameter multiple times and cycle through the specified filtergraphs along with the show modes by pressing the key w.

-af filtergraph

filtergraph is a description of the filtergraph to apply to the input audio. Use the option -filters to show all the available filters (including sources and sinks).

-i input_url

Read input_url.

-pix_fmt format

Set pixel format. This option has been deprecated in favor of private options, try -pixel_format.

-stats

Print several playback statistics, in particular show the stream duration, the codec parameters, the current position in the stream and the audio/video synchronisation drift. It is shown by default, unless the log level is lower than info. Its display can be forced by manually specifying this option. To disable it, you need to specify -nostats.

-fast

Non-spec-compliant optimizations.

-genpts

Generate pts.

-sync type

Set the master clock to audio (type=audio), video (type=video) or external (type=ext). Default is audio. The master clock is used to control audio-video synchronization. Most media players use audio as master clock, but in some cases (streaming or high quality broadcast) it is necessary to change that. This option is mainly used for debugging purposes.

-ast audio_stream_specifier

Select the desired audio stream using the given stream specifier. The stream specifiers are described in the Stream specifiers chapter. If this option is not specified, the best audio stream is selected in the program of the already selected video stream.

-vst video_stream_specifier

Select the desired video stream using the given stream specifier. The stream specifiers are described in the Stream specifiers chapter. If this option is not specified, the best video stream is selected.

-sst subtitle_stream_specifier

Select the desired subtitle stream using the given stream specifier. The stream specifiers are described in the Stream specifiers chapter. If this option is not specified, the best subtitle stream is selected in the program of the already selected video or audio stream.

-autoexit

Exit when video is done playing.

-exitonkeydown

Exit if any key is pressed.

-exitonmousedown

Exit if any mouse button is pressed.

-codec:media_specifier codec_name

Force a specific decoder implementation for the stream identified by media_specifier, which can assume the values a (audio), v (video), and s subtitle.

-acodec codec_name

Force a specific audio decoder.

-vcodec codec_name

Force a specific video decoder.

-scodec codec_name

Force a specific subtitle decoder.

-autorotate

Automatically rotate the video according to file metadata. Enabled by default, use -noautorotate to disable it.

-framedrop

Drop video frames if video is out of sync. Enabled by default if the master clock is not set to video. Use this option to enable frame dropping for all master clock sources, use -noframedrop to disable it.

-infbuf

Do not limit the input buffer size, read as much data as possible from the input as soon as possible. Enabled by default for realtime streams, where data may be dropped if not read in time. Use this option to enable infinite buffers for all inputs, use -noinfbuf to disable it.

-filter_threads nb_threads

Defines how many threads are used to process a filter pipeline. Each pipeline will produce a thread pool with this many threads available for parallel processing. The default is 0 which means that the thread count will be determined by the number of available CPUs.

q, ESC

Quit.

(no term)

Toggle full screen.

p, SPC

Pause.

(no term)

Toggle mute.

9, 0

Decrease and increase volume respectively.

/, *

Decrease and increase volume respectively.

(no term)

Cycle audio channel in the current program.

(no term)

Cycle video channel.

(no term)

Cycle subtitle channel in the current program.

(no term)

Cycle program.

(no term)

Cycle video filters or show modes.

(no term)

Step to the next frame. Pause if the stream is not already paused, step to the next video frame, and pause.

left/right

Seek backward/forward 10 seconds.

down/up

Seek backward/forward 1 minute.

page down/page up

Seek to the previous/next chapter. or if there are no chapters Seek backward/forward 10 minutes.

right mouse click

Seek to percentage in file corresponding to fraction of width.

left mouse double-click

Toggle full screen.

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

The accepted syntax is:

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

If the value is now it takes the current time.

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

There are two accepted syntaxes for expressing time duration.

[-][<HH>:]<MM>:<SS>[.<m>…]

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

or

[-]<S>+[.<m>…][s|ms|us]

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

In both expressions, the optional - indicates negative duration.

Examples

The following examples are all valid time duration:

1. 55 seconds
2. 0.2 seconds
3. 200 milliseconds, that’s 0.2s
4. 200000 microseconds, that’s 0.2s
5. 12 hours, 03 minutes and 45 seconds
6. 23.189 seconds

Specify the size of the sourced video, it may be a string of the form width/x/height, or the name of a size abbreviation.

The following abbreviations are recognized:

ntsc

720x480

pal

720x576

qntsc

352x240

qpal

352x288

sntsc

640x480

spal

768x576

film

352x240

ntsc-film

352x240

sqcif

128x96

qcif

176x144

cif

352x288

4cif

704x576

16cif

1408x1152

qqvga

160x120

qvga

320x240

vga

640x480

svga

800x600

xga

1024x768

uxga

1600x1200

qxga

2048x1536

sxga

1280x1024

qsxga

2560x2048

hsxga

5120x4096

wvga

852x480

wxga

1366x768

wsxga

1600x1024

wuxga

1920x1200

woxga

2560x1600

wqsxga

3200x2048

wquxga

3840x2400

whsxga

6400x4096

whuxga

7680x4800

cga

320x200

ega

640x350

hd480

852x480

hd720

1280x720

hd1080

1920x1080

2k

2048x1080

2kflat

1998x1080

2kscope

2048x858

4k

4096x2160

4kflat

3996x2160

4kscope

4096x1716

nhd

640x360

hqvga

240x160

wqvga

400x240

fwqvga

432x240

hvga

480x320

qhd

960x540

2kdci

2048x1080

4kdci

4096x2160

uhd2160

3840x2160

uhd4320

7680x4320

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

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

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

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

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

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

The string random will result in a random color.

The following names of colors are recognized:

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

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

(no term)

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

(no term)

FL+FR+LFE

(no term)

FL+FR+FC

(no term)

FL+FR+BC

(no term)

FL+FR+FC+BC

quad

FL+FR+BL+BR

quad(side)

FL+FR+SL+SR

(no term)

FL+FR+FC+LFE

(no term)

FL+FR+FC+BL+BR

(no term)

FL+FR+FC+SL+SR

(no term)

FL+FR+FC+LFE+BC

(no term)

FL+FR+FC+LFE+BL+BR

(no term)

FL+FR+FC+LFE+SL+SR

(no term)

FL+FR+FC+BC+SL+SR

(no term)

FL+FR+FLC+FRC+SL+SR

hexagonal

FL+FR+FC+BL+BR+BC

(no term)

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

(no term)

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

(no term)

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

(no term)

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

(no term)

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

(no term)

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

(no term)

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

(no term)

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.

AOMedia Video 1 (AV1) decoder.

Options

operating_point

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

Raw video decoder.

This decoder decodes rawvideo streams.

Options

top top_field_first

Specify the assumed field type of the input video.

-1

the video is assumed to be progressive (default)

(no term)

bottom-field-first is assumed

(no term)

top-field-first is assumed

dav1d AV1 decoder.

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

Options

The following options are supported by the libdav1d wrapper.

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.

AVS2-P2/IEEE1857.4 video decoder wrapper.

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

AVS3-P2/IEEE1857.10 video decoder.

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

Options

The following option is supported by the libuavs3d wrapper.

frame_threads

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

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. The default value is 1. There are 3 notable scale factor ranges:

drc_scale == 0

DRC disabled. Produces full range audio.

0 < drc_scale <= 1

DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full range and full compression.

drc_scale > 1

DRC enabled. Applies drc_scale asymmetrically. Loud sounds are fully compressed. Soft sounds are enhanced.

FLAC 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.

Internal wave synthesizer.

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

libcelt decoder wrapper.

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

libgsm decoder wrapper.

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

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

libilbc decoder wrapper.

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

Options

The following option is supported by the libilbc wrapper.

enhance

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

libopencore-amrnb decoder wrapper.

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

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

libopencore-amrwb decoder wrapper.

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

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

libopus decoder wrapper.

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

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

ARIB STD-B24 caption decoder.

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

libaribb24 Decoder Options

-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.

Options

compute_clut

-1

Compute clut if no matching CLUT is in the stream.

1. Never compute CLUT
2. Always compute CLUT and override the one provided in the stream.

dvb_substream

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

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

Options

palette

Specify the global palette used by the bitmaps. When stored in VobSub, the palette is normally specified in the index file; in Matroska, the palette is stored in the codec extra-data in the same format as in VobSub. In DVDs, the palette is stored in the IFO file, and therefore not available when reading from dumped VOB files. The format for this option is a string containing 16 24-bits hexadecimal numbers (without 0x prefix) separated by commas, for example 0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b.

ifo_palette

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

forced_subs_only

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

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

Options

txt_page

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

txt_default_region

Set default character set used for decoding, a value between 0 and 87 (see ETS 300 706, Section 15, Table 32). Default value is -1, which does not override the libzvbi default. This option is needed for some legacy level 1.0 transmissions which cannot signal the proper charset.

txt_chop_top

Discards the top teletext line. Default value is 1.

txt_format

Specifies the format of the decoded subtitles.

bitmap

The default format, you should use this for teletext pages, because certain graphics and colors cannot be expressed in simple text or even ASS.

text

Simple text based output without formatting.

ass

Formatted ASS output, subtitle pages and teletext pages are returned in different styles, subtitle pages are stripped down to text, but an effort is made to keep the text alignment and the formatting.

txt_left

X offset of generated bitmaps, default is 0.

txt_top

Y offset of generated bitmaps, default is 0.

txt_chop_spaces

Chops leading and trailing spaces and removes empty lines from the generated text. This option is useful for teletext based subtitles where empty spaces may be present at the start or at the end of the lines or empty lines may be present between the subtitle lines because of double-sized teletext characters. Default value is 1.

txt_duration

Sets the display duration of the decoded teletext pages or subtitles in milliseconds. Default value is -1 which means infinity or until the next subtitle event comes.

txt_transparent

Force transparent background of the generated teletext bitmaps. Default value is 0 which means an opaque background.

txt_opacity

Sets the opacity (0-255) of the teletext background. If txt_transparent is not set, it only affects characters between a start box and an end box, typically subtitles. Default value is 0 if txt_transparent is set, 255 otherwise.

Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration bitstream.

This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS header and removes the ADTS header.

This filter is required for example when copying an AAC stream from a raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or to MOV/MP4 files and related formats such as 3GP or M4A. Please note that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.

Modify metadata embedded in an AV1 stream.

td

Insert or remove temporal delimiter OBUs in all temporal units of the stream.

insert

Insert a TD at the beginning of every TU which does not already have one.

remove

Remove the TD from the beginning of every TU which has one.

color_primaries

transfer_characteristics

matrix_coefficients

Set the color description fields in the stream (see AV1 section 6.4.2).

color_range

Set the color range in the stream (see AV1 section 6.4.2; note that this cannot be set for streams using BT.709 primaries, sRGB transfer characteristic and identity (RGB) matrix coefficients).

tv

Limited range.

pc

Full range.

chroma_sample_position

Set the chroma sample location in the stream (see AV1 section 6.4.2). This can only be set for 4:2:0 streams.

vertical

Left position (matching the default in MPEG-2 and H.264).

colocated

Top-left position.

tick_rate

Set the tick rate (num_units_in_display_tick / time_scale) in the timing info in the sequence header.

num_ticks_per_picture

Set the number of ticks in each picture, to indicate that the stream has a fixed framerate. Ignored if tick_rate is not also set.

delete_padding

Deletes Padding OBUs.

Remove zero padding at the end of a packet.

Extract the core from a DCA/DTS stream, dropping extensions such as DTS-HD.

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

freq

The additional argument specifies which packets should be filtered. It accepts the values:

add extradata to all key packets

add extradata to all packets

If not specified it is assumed k.

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

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

Extract the core from a E-AC-3 stream, dropping extra channels.

Extract the in-band extradata.

Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers, or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either in-band (i.e. as a part of the bitstream containing the coded frames) or out of band (e.g. on the container level). This latter form is called extradata in FFmpeg terminology.

This bitstream filter detects the in-band headers and makes them available as extradata.

remove

When this option is enabled, the long-term headers are removed from the bitstream after extraction.

Remove units with types in or not in a given set from the stream.

pass_types

List of unit types or ranges of unit types to pass through while removing all others. This is specified as a ’|’-separated list of unit type values or ranges of values with ’-’.

remove_types

Identical to pass_types, except the units in the given set removed and all others passed through.

Extradata is unchanged by this transformation, but note that if the stream contains inline parameter sets then the output may be unusable if they are removed.

For example, to remove all non-VCL NAL units from an H.264 stream:

ffmpeg -i INPUT -c:v copy -bsf:v filter_units=pass_types=1-5 OUTPUT

To remove all AUDs, SEI and filler from an H.265 stream:

ffmpeg -i INPUT -c:v copy -bsf:v filter_units=remove_types=35|38-40 OUTPUT

Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.

texture

Specifies the texture to keep.

color

alpha

Convert HAPQA to HAPQ

ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder=“HAPQ” hapq_file.mov

Convert HAPQA to HAPAlphaOnly

ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder=“HAPAlpha Only” hapalphaonly_file.mov

Modify metadata embedded in an H.264 stream.

aud

Insert or remove AUD NAL units in all access units of the stream.

insert

remove

sample_aspect_ratio

Set the sample aspect ratio of the stream in the VUI parameters.

overscan_appropriate_flag

Set whether the stream is suitable for display using overscan or not (see H.264 section E.2.1).

video_format

video_full_range_flag

Set the video format in the stream (see H.264 section E.2.1 and table E-2).

colour_primaries

transfer_characteristics

matrix_coefficients

Set the colour description in the stream (see H.264 section E.2.1 and tables E-3, E-4 and E-5).

chroma_sample_loc_type

Set the chroma sample location in the stream (see H.264 section E.2.1 and figure E-1).

tick_rate

Set the tick rate (num_units_in_tick / time_scale) in the VUI parameters. This is the smallest time unit representable in the stream, and in many cases represents the field rate of the stream (double the frame rate).

fixed_frame_rate_flag

Set whether the stream has fixed framerate - typically this indicates that the framerate is exactly half the tick rate, but the exact meaning is dependent on interlacing and the picture structure (see H.264 section E.2.1 and table E-6).

crop_left

crop_right

crop_top

crop_bottom

Set the frame cropping offsets in the SPS. These values will replace the current ones if the stream is already cropped. These fields are set in pixels. Note that some sizes may not be representable if the chroma is subsampled or the stream is interlaced (see H.264 section 7.4.2.1.1).

sei_user_data

Insert a string as SEI unregistered user data. The argument must be of the form UUID+string, where the UUID is as hex digits possibly separated by hyphens, and the string can be anything. For example, 086f3693-b7b3-4f2c-9653-21492feee5b8+hello will insert the string ``hello’’ associated with the given UUID.

delete_filler

Deletes both filler NAL units and filler SEI messages.

level

Set the level in the SPS. Refer to H.264 section A.3 and tables A-1 to A-5. The argument must be the name of a level (for example, 4.2), a level_idc value (for example, 42), or the special name auto indicating that the filter should attempt to guess the level from the input stream properties.

Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.264 specification).

This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer mpegts).

For example to remux an MP4 file containing an H.264 stream to mpegts format with ffmpeg, you can use the command:

ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts

Please note that this filter is auto-inserted for MPEG-TS (muxer mpegts) and raw H.264 (muxer h264) output formats.

This applies a specific fixup to some Blu-ray streams which contain redundant PPSs modifying irrelevant parameters of the stream which confuse other transformations which require correct extradata.

A new single global PPS is created, and all of the redundant PPSs within the stream are removed.

Modify metadata embedded in an HEVC stream.

aud

Insert or remove AUD NAL units in all access units of the stream.

insert

remove

sample_aspect_ratio

Set the sample aspect ratio in the stream in the VUI parameters.

video_format

video_full_range_flag

Set the video format in the stream (see H.265 section E.3.1 and table E.2).

colour_primaries

transfer_characteristics

matrix_coefficients

Set the colour description in the stream (see H.265 section E.3.1 and tables E.3, E.4 and E.5).

chroma_sample_loc_type

Set the chroma sample location in the stream (see H.265 section E.3.1 and figure E.1).

tick_rate

Set the tick rate in the VPS and VUI parameters (num_units_in_tick / time_scale). Combined with num_ticks_poc_diff_one, this can set a constant framerate in the stream. Note that it is likely to be overridden by container parameters when the stream is in a container.

num_ticks_poc_diff_one

Set poc_proportional_to_timing_flag in VPS and VUI and use this value to set num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and E.3.1). Ignored if tick_rate is not also set.

crop_left

crop_right

crop_top

crop_bottom

Set the conformance window cropping offsets in the SPS. These values will replace the current ones if the stream is already cropped. These fields are set in pixels. Note that some sizes may not be representable if the chroma is subsampled (H.265 section 7.4.3.2.1).

level

Set the level in the VPS and SPS. See H.265 section A.4 and tables A.6 and A.7. The argument must be the name of a level (for example, 5.1), a general_level_idc value (for example, 153 for level 5.1), or the special name auto indicating that the filter should attempt to guess the level from the input stream properties.

Convert an HEVC/H.265 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.265 specification).

This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer mpegts).

For example to remux an MP4 file containing an HEVC stream to mpegts format with ffmpeg, you can use the command:

ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts

Please note that this filter is auto-inserted for MPEG-TS (muxer mpegts) and raw HEVC/H.265 (muxer h265 or hevc) output formats.

Modifies the bitstream to fit in MOV and to be usable by the Final Cut Pro decoder. This filter only applies to the mpeg2video codec, and is likely not needed for Final Cut Pro 7 and newer with the appropriate -tag:v.

For example, to remux 30 MB/sec NTSC IMX to MOV:

ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov

Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.

MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be extracted without loss, e.g. by

ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg

Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for decoding. Quoting from <*http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml*>:

Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that MJPEG, or at least the MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed Ω- and omitted Ω- Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won’t have any idea how to decompress the data. The exact table necessary is given in the OpenDML spec.

This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images.

ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg exiftran -i -9 frame*.jpg ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

Add an MJPEG A header to the bitstream, to enable decoding by Quicktime.

Extract a representable text file from MOV subtitles, stripping the metadata header from each subtitle packet.

See also the text2movsub filter.

Decompress non-standard compressed MP3 audio headers.

Modify metadata embedded in an MPEG-2 stream.

display_aspect_ratio

Set the display aspect ratio in the stream. The following fixed values are supported:

4/3

16/9

221/100

Any other value will result in square pixels being signalled instead (see H.262 section 6.3.3 and table 6-3).

frame_rate

Set the frame rate in the stream. This is constructed from a table of known values combined with a small multiplier and divisor - if the supplied value is not exactly representable, the nearest representable value will be used instead (see H.262 section 6.3.3 and table 6-4).

video_format

Set the video format in the stream (see H.262 section 6.3.6 and table 6-6).

colour_primaries

transfer_characteristics

matrix_coefficients

Set the colour description in the stream (see H.262 section 6.3.6 and tables 6-7, 6-8 and 6-9).

Unpack DivX-style packed B-frames.

DivX-style packed B-frames are not valid MPEG-4 and were only a workaround for the broken Video for Windows subsystem. They use more space, can cause minor AV sync issues, require more CPU power to decode (unless the player has some decoded picture queue to compensate the 2,0,2,0 frame per packet style) and cause trouble if copied into a standard container like mp4 or mpeg-ps/ts, because MPEG-4 decoders may not be able to decode them, since they are not valid MPEG-4.

For example to fix an AVI file containing an MPEG-4 stream with DivX-style packed B-frames using ffmpeg, you can use the command:

ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi

Damages the contents of packets or simply drops them without damaging the container. Can be used for fuzzing or testing error resilience/concealment.

Parameters:

amount

A numeral string, whose value is related to how often output bytes will be modified. Therefore, values below or equal to 0 are forbidden, and the lower the more frequent bytes will be modified, with 1 meaning every byte is modified.

dropamount

A numeral string, whose value is related to how often packets will be dropped. Therefore, values below or equal to 0 are forbidden, and the lower the more frequent packets will be dropped, with 1 meaning every packet is dropped.

The following example applies the modification to every byte but does not drop any packets.

ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv

This bitstream filter passes the packets through unchanged.

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

nb_out_samples, n

Set the number of samples per each output audio packet. The number is intended as the number of samples per each channel. Default value is 1024.

pad, p

If set to 1, the filter will pad the last audio packet with silence, so that it will contain the same number of samples (or roughly the same number of samples, see frame_rate) as the previous ones. Default value is 1.

frame_rate, r

This option makes the filter output a fixed number of packets per second instead of a fixed number of samples per packet. If the audio sample rate is not divisible by the frame rate then the number of samples will not be constant but will vary slightly so that each packet will start as close to the frame boundary as possible. Using this option has precedence over nb_out_samples.

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

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

Modify color property metadata embedded in prores stream.

color_primaries

Set the color primaries. Available values are:

auto

Keep the same color primaries property (default).

unknown

bt709

bt470bg

BT601 625

smpte170m

BT601 525

bt2020

smpte431

DCI P3

smpte432

P3 D65

transfer_characteristics

Set the color transfer. Available values are:

auto

Keep the same transfer characteristics property (default).

unknown

bt709

BT 601, BT 709, BT 2020

smpte2084

SMPTE ST 2084

arib-std-b67

ARIB STD-B67

matrix_coefficients

Set the matrix coefficient. Available values are:

auto

Keep the same colorspace property (default).

unknown

bt709

smpte170m

BT 601

bt2020nc

Set Rec709 colorspace for each frame of the file

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

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

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

Remove extradata from packets.

It accepts the following parameter:

freq

Set which frame types to remove extradata from.

1. Remove extradata from non-keyframes only.
2. Remove extradata from keyframes only.
3. Remove extradata from all frames.

Set PTS and DTS in packets.

It accepts the following parameters:

ts

pts

dts

Set expressions for PTS, DTS or both.

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

1. The count of the input packet. Starting from 0.
2. The demux timestamp in input in case of ts or dts option or presentation timestamp in case of pts option.
3. The original position in the file of the packet, or undefined if undefined for the current packet
4. The demux timestamp in input.
5. The presentation timestamp in input.
6. The DTS of the first packet.
7. The PTS of the first packet.
8. The previous input DTS.
9. The previous input PTS.
10. The previous output DTS.
11. The previous output PTS.
12. The timebase of stream packet belongs.
13. The sample rate of stream packet belongs.

Convert text subtitles to MOV subtitles (as used by the mov_text codec) with metadata headers.

See also the mov2textsub filter.

Log trace output containing all syntax elements in the coded stream headers (everything above the level of individual coded blocks). This can be useful for debugging low-level stream issues.

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

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

Modify metadata embedded in a VP9 stream.

color_space

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

unknown

bt601

bt709

smpte170

smpte240

bt2020

rgb

color_range

Set the color range value in the frame header. Note that any value imposed by the color space will take precedence over this value.

tv

pc

nil

Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This fixes merging of split/segmented VP9 streams where the alt-ref frame was split from its visible counterpart.

Split VP9 superframes into single frames.

Given a VP9 stream with correct timestamps but possibly out of order, insert additional show-existing-frame packets to correct the ordering.

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

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

Audible Format 2, 3, and 4 demuxer.

This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.

Animated Portable Network Graphics demuxer.

This demuxer is used to demux APNG files. All headers, but the PNG signature, up to (but not including) the first fcTL chunk are transmitted as extradata. Frames are then split as being all the chunks between two fcTL ones, or between the last fcTL and IEND chunks.

-ignore_loop bool

Ignore the loop variable in the file if set.

-max_fps int

Maximum framerate in frames per second (0 for no limit).

-default_fps int

Default framerate in frames per second when none is specified in the file (0 meaning as fast as possible).

Advanced Systems Format demuxer.

This demuxer is used to demux ASF files and MMS network streams.

-no_resync_search bool

Do not try to resynchronize by looking for a certain optional start code.

Virtual concatenation script demuxer.

This demuxer reads a list of files and other directives from a text file and demuxes them one after the other, as if all their packets had been muxed together.

The timestamps in the files are adjusted so that the first file starts at 0 and each next file starts where the previous one finishes. Note that it is done globally and may cause gaps if all streams do not have exactly the same length.

All files must have the same streams (same codecs, same time base, etc.).

The duration of each file is used to adjust the timestamps of the next file: if the duration is incorrect (because it was computed using the bit-rate or because the file is truncated, for example), it can cause artifacts. The duration directive can be used to override the duration stored in each file.

Syntax

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

“file path”

Path to a file to read; special characters and spaces must be escaped with backslash or single quotes. All subsequent file-related directives apply to that file.

“ffconcat version 1.0”

Identify the script type and version. It also sets the safe option to 1 if it was -1. To make FFmpeg recognize the format automatically, this directive must appear exactly as is (no extra space or byte-order-mark) on the very first line of the script.

“duration dur”

Duration of the file. This information can be specified from the file; specifying it here may be more efficient or help if the information from the file is not available or accurate. If the duration is set for all files, then it is possible to seek in the whole concatenated video.

“inpoint timestamp”

In point of the file. When the demuxer opens the file it instantly seeks to the specified timestamp. Seeking is done so that all streams can be presented successfully at In point. This directive works best with intra frame codecs, because for non-intra frame ones you will usually get extra packets before the actual In point and the decoded content will most likely contain frames before In point too. For each file, packets before the file In point will have timestamps less than the calculated start timestamp of the file (negative in case of the first file), and the duration of the files (if not specified by the duration directive) will be reduced based on their specified In point. Because of potential packets before the specified In point, packet timestamps may overlap between two concatenated files.

“outpoint timestamp”

Out point of the file. When the demuxer reaches the specified decoding timestamp in any of the streams, it handles it as an end of file condition and skips the current and all the remaining packets from all streams. Out point is exclusive, which means that the demuxer will not output packets with a decoding timestamp greater or equal to Out point. This directive works best with intra frame codecs and formats where all streams are tightly interleaved. For non-intra frame codecs you will usually get additional packets with presentation timestamp after Out point therefore the decoded content will most likely contain frames after Out point too. If your streams are not tightly interleaved you may not get all the packets from all streams before Out point and you may only will be able to decode the earliest stream until Out point. The duration of the files (if not specified by the duration directive) will be reduced based on their specified Out point.

“file_packet_metadata key=value”

Metadata of the packets of the file. The specified metadata will be set for each file packet. You can specify this directive multiple times to add multiple metadata entries.

“stream”

Introduce a stream in the virtual file. All subsequent stream-related directives apply to the last introduced stream. Some streams properties must be set in order to allow identifying the matching streams in the subfiles. If no streams are defined in the script, the streams from the first file are copied.

“exact_stream_id id”

Set the id of the stream. If this directive is given, the string with the corresponding id in the subfiles will be used. This is especially useful for MPEG-PS (VOB) files, where the order of the streams is not reliable.

Options

This demuxer accepts the following option:

safe

If set to 1, reject unsafe file paths. A file path is considered safe if it does not contain a protocol specification and is relative and all components only contain characters from the portable character set (letters, digits, period, underscore and hyphen) and have no period at the beginning of a component. If set to 0, any file name is accepted. The default is 1. -1 is equivalent to 1 if the format was automatically probed and 0 otherwise.

auto_convert

If set to 1, try to perform automatic conversions on packet data to make the streams concatenable. The default is 1. Currently, the only conversion is adding the h264_mp4toannexb bitstream filter to H.264 streams in MP4 format. This is necessary in particular if there are resolution changes.

segment_time_metadata

If set to 1, every packet will contain the lavf.concat.start_time and the lavf.concat.duration packet metadata values which are the start_time and the duration of the respective file segments in the concatenated output expressed in microseconds. The duration metadata is only set if it is known based on the concat file. The default is 0.

Examples

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

Dynamic Adaptive Streaming over HTTP demuxer.

This demuxer presents all AVStreams found in the manifest. By setting the discard flags on AVStreams the caller can decide which streams to actually receive. Each stream mirrors the id and bandwidth properties from the <Representation> as metadata keys named id and variant_bitrate respectively.

Adobe Flash Video Format demuxer.

This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities.

ffmpeg -f flv -i myfile.flv … ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ….

-flv_metadata bool

Allocate the streams according to the onMetaData array content.

-flv_ignore_prevtag bool

Ignore the size of previous tag value.

-flv_full_metadata bool

Output all context of the onMetadata.

Animated GIF demuxer.

It accepts the following options:

min_delay

Set the minimum valid delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 2.

max_gif_delay

Set the maximum valid delay between frames in hundredth of seconds. Range is 0 to 65535. Default value is 65535 (nearly eleven minutes), the maximum value allowed by the specification.

default_delay

Set the default delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 10.

ignore_loop

GIF files can contain information to loop a certain number of times (or infinitely). If ignore_loop is set to 1, then the loop setting from the input will be ignored and looping will not occur. If set to 0, then looping will occur and will cycle the number of times according to the GIF. Default value is 1.

For example, with the overlay filter, place an infinitely looping GIF over another video:

ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv

Note that in the above example the shortest option for overlay filter is used to end the output video at the length of the shortest input file, which in this case is input.mp4 as the GIF in this example loops infinitely.

HLS demuxer

Apple HTTP Live Streaming demuxer.

This demuxer presents all AVStreams from all variant streams. The id field is set to the bitrate variant index number. By setting the discard flags on AVStreams (by pressing ’a’ or ’v’ in ffplay), the caller can decide which variant streams to actually receive. The total bitrate of the variant that the stream belongs to is available in a metadata key named variant_bitrate.

It accepts the following options:

live_start_index

segment index to start live streams at (negative values are from the end).

allowed_extensions

’,’ separated list of file extensions that hls is allowed to access.

max_reload

Maximum number of times a insufficient list is attempted to be reloaded. Default value is 1000.

m3u8_hold_counters

The maximum number of times to load m3u8 when it refreshes without new segments. Default value is 1000.

http_persistent

Use persistent HTTP connections. Applicable only for HTTP streams. Enabled by default.

http_multiple

Use multiple HTTP connections for downloading HTTP segments. Enabled by default for HTTP/1.1 servers.

http_seekable

Use HTTP partial requests for downloading HTTP segments. 0 = disable, 1 = enable, -1 = auto, Default is auto.

Image file demuxer.

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

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

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

This demuxer accepts the following options:

framerate

Set the frame rate for the video stream. It defaults to 25.

loop

If set to 1, loop over the input. Default value is 0.

pattern_type

Select the pattern type used to interpret the provided filename. pattern_type accepts one of the following values.

none

Disable pattern matching, therefore the video will only contain the specified image. You should use this option if you do not want to create sequences from multiple images and your filenames may contain special pattern characters.

sequence

Select a sequence pattern type, used to specify a sequence of files indexed by sequential numbers. A sequence pattern may contain the string %d or “%0/N/d, which specifies the position of the characters representing a sequential number in each filename matched by the pattern. If the form %d0/N/d” is used, the string representing the number in each filename is 0-padded and N is the total number of 0-padded digits representing the number. The literal character ’%’ can be specified in the pattern with the string %%. If the sequence pattern contains %d or “%0/N/d”, the first filename of the file list specified by the pattern must contain a number inclusively contained between start_number and start_number/+/start_number_range-1, and all the following numbers must be sequential. For example the pattern img-%03d.bmp will match a sequence of filenames of the form img-001.bmp, img-002.bmp, …, img-010.bmp, etc.; the pattern i%%m%%g-%d.jpg will match a sequence of filenames of the form i%m%g-1.jpg, i%m%g-2.jpg, …, i%m%g-10.jpg, etc. Note that the pattern must not necessarily contain %d or “%0/N/d”, for example to convert a single image file img.jpeg you can employ the command: ffmpeg -i img.jpeg img.png

glob

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

glob_sequence (deprecated, will be removed)

Select a mixed glob wildcard/sequence pattern. If your version of libavformat was compiled with globbing support, and the provided pattern contains at least one glob meta character among %*?[]{} that is preceded by an unescaped %, the pattern is interpreted like a glob() pattern, otherwise it is interpreted like a sequence pattern. All glob special characters %*?[]{} must be prefixed with %. To escape a literal % you shall use %%. For example the pattern foo-%*.jpeg will match all the filenames prefixed by foo- and terminating with .jpeg, and foo-%?%?%?.jpeg will match all the filenames prefixed with foo-, followed by a sequence of three characters, and terminating with .jpeg. This pattern type is deprecated in favor of glob and sequence.

Default value is glob_sequence.

pixel_format

Set the pixel format of the images to read. If not specified the pixel format is guessed from the first image file in the sequence.

start_number

Set the index of the file matched by the image file pattern to start to read from. Default value is 0.

start_number_range

Set the index interval range to check when looking for the first image file in the sequence, starting from start_number. Default value is 5.

ts_from_file

If set to 1, will set frame timestamp to modification time of image file. Note that monotonity of timestamps is not provided: images go in the same order as without this option. Default value is 0. If set to 2, will set frame timestamp to the modification time of the image file in nanosecond precision.

video_size

Set the video size of the images to read. If not specified the video size is guessed from the first image file in the sequence.

export_path_metadata

If set to 1, will add two extra fields to the metadata found in input, making them also available for other filters (see drawtext filter for examples). Default value is 0. The extra fields are described below:

lavf.image2dec.source_path

Corresponds to the full path to the input file being read.

lavf.image2dec.source_basename

Corresponds to the name of the file being read.

Examples

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

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

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

It accepts the following options:

track_index

Set the index of which track to demux. The demuxer can only export one track. Track indexes start at 0. Default is to pick the first track. Number of tracks is exported as tracks metadata entry.

sample_rate

Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.

max_size (bytes)

The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of files that can be read. Default is 50 MiB.

ModPlug based module demuxer

See <*https://github.com/Konstanty/libmodplug*>

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

It accepts the following options:

noise_reduction

Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.

reverb_depth

Set amount of reverb. Range 0-100. Default is 0.

reverb_delay

Set delay in ms, clamped to 40-250 ms. Default is 0.

bass_amount

Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.

bass_range

Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.

surround_depth

Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.

surround_delay

Set surround delay in ms, clamped to 5-40 ms. Default is 0.

max_size

The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB. 0 removes buffer size limit (not recommended). Default is 5 MiB.

video_stream_expr

String which is evaluated using the eval API to assign colors to the generated video stream. Variables which can be used are x, y, w, h, t, speed, tempo, order, pattern and row.

video_stream

Generate video stream. Can be 1 (on) or 0 (off). Default is 0.

video_stream_w

Set video frame width in ’chars’ where one char indicates 8 pixels. Range is 20-512. Default is 30.

video_stream_h

Set video frame height in ’chars’ where one char indicates 8 pixels. Range is 20-512. Default is 30.

video_stream_ptxt

Print metadata on video stream. Includes speed, tempo, order, pattern, row and ts (time in ms). Can be 1 (on) or 0 (off). Default is 1.

libopenmpt based module demuxer

See <*https://lib.openmpt.org/libopenmpt/*> for more information.

Some files have multiple subsongs (tracks) this can be set with the subsong option.

It accepts the following options:

subsong

Set the subsong index. This can be either ’all’, ’auto’, or the index of the subsong. Subsong indexes start at 0. The default is ’auto’. The default value is to let libopenmpt choose.

layout

Set the channel layout. Valid values are 1, 2, and 4 channel layouts. The default value is STEREO.

sample_rate

Set the sample rate for libopenmpt to output. Range is from 1000 to INT_MAX. The value default is 48000.

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

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

Options

This demuxer accepts the following options:

enable_drefs

Enable loading of external tracks, disabled by default. Enabling this can theoretically leak information in some use cases.

use_absolute_path

Allows loading of external tracks via absolute paths, disabled by default. Enabling this poses a security risk. It should only be enabled if the source is known to be non-malicious.

seek_streams_individually

When seeking, identify the closest point in each stream individually and demux packets in that stream from identified point. This can lead to a different sequence of packets compared to demuxing linearly from the beginning. Default is true.

ignore_editlist

Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the timeline described by the edit list. Default is false.

advanced_editlist

Modify the stream index to reflect the timeline described by the edit list. ignore_editlist must be set to false for this option to be effective. If both ignore_editlist and this option are set to false, then only the start of the stream index is modified to reflect initial dwell time or starting timestamp described by the edit list. Default is true.

ignore_chapters

Don’t parse chapters. This includes GoPro ’HiLight’ tags/moments. Note that chapters are only parsed when input is seekable. Default is false.

use_mfra_for

For seekable fragmented input, set fragment’s starting timestamp from media fragment random access box, if present. Following options are available:

auto

Auto-detect whether to set mfra timestamps as PTS or DTS (default)

dts

Set mfra timestamps as DTS

pts

Set mfra timestamps as PTS

(no term)

Don’t use mfra box to set timestamps

export_all

Export unrecognized boxes within the udta box as metadata entries. The first four characters of the box type are set as the key. Default is false.

export_xmp

Export entire contents of XMP_ box and uuid box as a string with key xmp. Note that if export_all is set and this option isn’t, the contents of XMP_ box are still exported but with key XMP_. Default is false.

activation_bytes

4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.

audible_fixed_key

Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to specify.

decryption_key

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

Audible AAX

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

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

MPEG-2 transport stream demuxer.

This demuxer accepts the following options:

resync_size

Set size limit for looking up a new synchronization. Default value is 65536.

skip_unknown_pmt

Skip PMTs for programs not defined in the PAT. Default value is 0.

fix_teletext_pts

Override teletext packet PTS and DTS values with the timestamps calculated from the PCR of the first program which the teletext stream is part of and is not discarded. Default value is 1, set this option to 0 if you want your teletext packet PTS and DTS values untouched.

ts_packetsize

Output option carrying the raw packet size in bytes. Show the detected raw packet size, cannot be set by the user.

scan_all_pmts

Scan and combine all PMTs. The value is an integer with value from -1 to 1 (-1 means automatic setting, 1 means enabled, 0 means disabled). Default value is -1.

merge_pmt_versions

Re-use existing streams when a PMT’s version is updated and elementary streams move to different PIDs. Default value is 0.

MJPEG encapsulated in multi-part MIME demuxer.

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

strict_mime_boundary

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

Raw video demuxer.

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

This demuxer accepts the following options:

framerate

Set input video frame rate. Default value is 25.

pixel_format

Set the input video pixel format. Default value is yuv420p.

video_size

Set the input video size. This value must be specified explicitly.

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

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

SBaGen script demuxer.

This demuxer reads the script language used by SBaGen <*http://uazu.net/sbagen/*> to generate binaural beats sessions. A SBG script looks like that:

-SE a: 300-2.5/3 440+4.5/0 b: 300-2.5/0 440+4.5/3 off: - NOW = a +0:07:00 = b +0:14:00 = a +0:21:00 = b +0:30:00 off

A SBG script can mix absolute and relative timestamps. If the script uses either only absolute timestamps (including the script start time) or only relative ones, then its layout is fixed, and the conversion is straightforward. On the other hand, if the script mixes both kind of timestamps, then the NOW reference for relative timestamps will be taken from the current time of day at the time the script is read, and the script layout will be frozen according to that reference. That means that if the script is directly played, the actual times will match the absolute timestamps up to the sound controller’s clock accuracy, but if the user somehow pauses the playback or seeks, all times will be shifted accordingly.

JSON captions used for <*http://www.ted.com/*>.

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

This demuxer accepts the following option:

start_time

Set the start time of the TED talk, in milliseconds. The default is 15000 (15s). It is used to sync the captions with the downloadable videos, because they include a 15s intro.

Example: convert the captions to a format most players understand:

ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt

Vapoursynth wrapper.

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

This demuxer accepts the following option:

max_script_size

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

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

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

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

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

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

Muliple subscribers may stream from the broker using the command:

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

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

The following options are supported:

exchange

Sets the exchange to use on the broker. RabbitMQ has several predefined exchanges: amq.direct is the default exchange, where the publisher and subscriber must have a matching routing_key; amq.fanout is the same as a broadcast operation (i.e. the data is forwarded to all queues on the fanout exchange independent of the routing_key); and amq.topic is similar to amq.direct, but allows for more complex pattern matching (refer to the RabbitMQ documentation).

routing_key

Sets the routing key. The default value is amqp. The routing key is used on the amq.direct and amq.topic exchanges to decide whether packets are written to the queue of a subscriber.

pkt_size

Maximum size of each packet sent/received to the broker. Default is 131072. Minimum is 4096 and max is any large value (representable by an int). When receiving packets, this sets an internal buffer size in FFmpeg. It should be equal to or greater than the size of the published packets to the broker. Otherwise the received message may be truncated causing decoding errors.

connection_timeout

The timeout in seconds during the initial connection to the broker. The default value is rw_timeout, or 5 seconds if rw_timeout is not set.

delivery_mode mode

Sets the delivery mode of each message sent to broker. The following values are accepted:

persistent

Delivery mode set to persistent (2). This is the default value. Messages may be written to the broker’s disk depending on its setup.

non-persistent

Delivery mode set to non-persistent (1). Messages will stay in broker’s memory unless the broker is under memory pressure.

Asynchronous data filling wrapper for input stream.

Fill data in a background thread, to decouple I/O operation from demux thread.

async:<URL> async:http://host/resource async:cache:http://host/resource

Read BluRay playlist.

The accepted options are:

angle

BluRay angle

chapter

Start chapter (1…N)

playlist

Playlist to read (BDMV/PLAYLIST/?????.mpls)

Examples:

Read longest playlist from BluRay mounted to /mnt/bluray:

bluray:/mnt/bluray

Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:

-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray

Caching wrapper for input stream.

Cache the input stream to temporary file. It brings seeking capability to live streams.

The accepted options are:

read_ahead_limit

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

URL Syntax is

cache:<URL>

Physical concatenation protocol.

Read and seek from many resources in sequence as if they were a unique resource.

A URL accepted by this protocol has the syntax:

concat:<URL1>|<URL2>|…|<URLN>

where URL1, URL2, …, URLN are the urls of the resource to be concatenated, each one possibly specifying a distinct protocol.

For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with ffplay use the command:

ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

Note that you may need to escape the character | which is special for many shells.

AES-encrypted stream reading protocol.

The accepted options are:

key

Set the AES decryption key binary block from given hexadecimal representation.

(no term)

Set the AES decryption initialization vector binary block from given hexadecimal representation.

Accepted URL formats:

crypto:<URL> crypto+<URL>

Data in-line in the URI. See <*http://en.wikipedia.org/wiki/Data_URI_scheme*>.

For example, to convert a GIF file given inline with ffmpeg:

ffmpeg -i “data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=” smiley.png

File access protocol.

Read from or write to a file.

A file URL can have the form:

file:<filename>

where filename is the path of the file to read.

An URL that does not have a protocol prefix will be assumed to be a file URL. Depending on the build, an URL that looks like a Windows path with the drive letter at the beginning will also be assumed to be a file URL (usually not the case in builds for unix-like systems).

For example to read from a file input.mpeg with ffmpeg use the command:

ffmpeg -i input.mpeg output.mpeg

This protocol accepts the following options:

truncate

Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.

blocksize

Set I/O operation maximum block size, in bytes. Default value is INT_MAX, which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable for files on slow medium.

follow

If set to 1, the protocol will retry reading at the end of the file, allowing reading files that still are being written. In order for this to terminate, you either need to use the rw_timeout option, or use the interrupt callback (for API users).

seekable

Controls if seekability is advertised on the file. 0 means non-seekable, -1 means auto (seekable for normal files, non-seekable for named pipes). Many demuxers handle seekable and non-seekable resources differently, overriding this might speed up opening certain files at the cost of losing some features (e.g. accurate seeking).

FTP (File Transfer Protocol).

Read from or write to remote resources using FTP protocol.

Following syntax is required.

ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

This protocol accepts the following options.

timeout

Set timeout in microseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified.

ftp-user

Set a user to be used for authenticating to the FTP server. This is overridden by the user in the FTP URL.

ftp-password

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

ftp-anonymous-password

Password used when login as anonymous user. Typically an e-mail address should be used.

ftp-write-seekable

Control seekability of connection during encoding. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable. Default value is 0.

NOTE: Protocol can be used as output, but it is recommended to not do it, unless special care is taken (tests, customized server configuration etc.). Different FTP servers behave in different way during seek operation. ff* tools may produce incomplete content due to server limitations.

Gopher protocol.

Gophers protocol.

The Gopher protocol with TLS encapsulation.

Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying “+/proto/” after the hls URI scheme name, where proto is either file or http.

hls+http://host/path/to/remote/resource.m3u8 hls+file://path/to/local/resource.m3u8

Using this protocol is discouraged - the hls demuxer should work just as well (if not, please report the issues) and is more complete. To use the hls demuxer instead, simply use the direct URLs to the m3u8 files.

HTTP (Hyper Text Transfer Protocol).

This protocol accepts the following options:

seekable

Control seekability of connection. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable, if set to -1 it will try to autodetect if it is seekable. Default value is -1.

chunked_post

If set to 1 use chunked Transfer-Encoding for posts, default is 1.

content_type

Set a specific content type for the POST messages or for listen mode.

http_proxy

set HTTP proxy to tunnel through e.g. http://example.com:1234

headers

Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers.

multiple_requests

Use persistent connections if set to 1, default is 0.

post_data

Set custom HTTP post data.

referer

Set the Referer header. Include ’Referer: URL’ header in HTTP request.

user_agent

Override the User-Agent header. If not specified the protocol will use a string describing the libavformat build. (Lavf/<version>)

user-agent

This is a deprecated option, you can use user_agent instead it.

reconnect_at_eof

If set then eof is treated like an error and causes reconnection, this is useful for live / endless streams.

reconnect_streamed

If set then even streamed/non seekable streams will be reconnected on errors.

reconnect_on_network_error

Reconnect automatically in case of TCP/TLS errors during connect.

reconnect_on_http_error

A comma separated list of HTTP status codes to reconnect on. The list can include specific status codes (e.g. ’503’) or the strings ’4xx’ / ’5xx’.

reconnect_delay_max

Sets the maximum delay in seconds after which to give up reconnecting

mime_type

Export the MIME type.

http_version

Exports the HTTP response version number. Usually 1.0 or 1.1.

icy

If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports this, the metadata has to be retrieved by the application by reading the icy_metadata_headers and icy_metadata_packet options. The default is 1.

icy_metadata_headers

If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers, separated by newline characters.

icy_metadata_packet

If the server supports ICY metadata, and icy was set to 1, this contains the last non-empty metadata packet sent by the server. It should be polled in regular intervals by applications interested in mid-stream metadata updates.

cookies

Set the cookies to be sent in future requests. The format of each cookie is the same as the value of a Set-Cookie HTTP response field. Multiple cookies can be delimited by a newline character.

offset

Set initial byte offset.

end_offset

Try to limit the request to bytes preceding this offset.

method

When used as a client option it sets the HTTP method for the request. When used as a server option it sets the HTTP method that is going to be expected from the client(s). If the expected and the received HTTP method do not match the client will be given a Bad Request response. When unset the HTTP method is not checked for now. This will be replaced by autodetection in the future.

listen

If set to 1 enables experimental HTTP server. This can be used to send data when used as an output option, or read data from a client with HTTP POST when used as an input option. If set to 2 enables experimental multi-client HTTP server. This is not yet implemented in ffmpeg.c and thus must not be used as a command line option. # Server side (sending): ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port> # Client side (receiving): ffmpeg -i http://<server>:<port> -c copy somefile.ogg # Client can also be done with wget: wget http://<server>:<port> -O somefile.ogg # Server side (receiving): ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg # Client side (sending): ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port> # Client can also be done with wget: wget –post-file=somefile.ogg http://<server>:<port>

send_expect_100

Send an Expect: 100-continue header for POST. If set to 1 it will send, if set to 0 it won’t, if set to -1 it will try to send if it is applicable. Default value is -1.

auth_type

Set HTTP authentication type. No option for Digest, since this method requires getting nonce parameters from the server first and can’t be used straight away like Basic.

none

Choose the HTTP authentication type automatically. This is the default.

basic

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

HTTP Cookies

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

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

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

Icecast protocol (stream to Icecast servers)

This protocol accepts the following options:

ice_genre

Set the stream genre.

ice_name

Set the stream name.

ice_description

Set the stream description.

ice_url

Set the stream website URL.

ice_public

Set if the stream should be public. The default is 0 (not public).

user_agent

Override the User-Agent header. If not specified a string of the form Lavf/<version> will be used.

password

Set the Icecast mountpoint password.

content_type

Set the stream content type. This must be set if it is different from audio/mpeg.

legacy_icecast

This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT method but the SOURCE method.

tls

Establish a TLS (HTTPS) connection to Icecast.

icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>

MMS (Microsoft Media Server) protocol over TCP.

MMS (Microsoft Media Server) protocol over HTTP.

The required syntax is:

mmsh://<server>[:<port>][/<app>][/<playpath>]

MD5 output protocol.

Computes the MD5 hash of the data to be written, and on close writes this to the designated output or stdout if none is specified. It can be used to test muxers without writing an actual file.

Some examples follow.

ffmpeg -i input.flv -f avi -y md5:output.avi.md5 # Write the MD5 hash of the encoded AVI file to stdout. ffmpeg -i input.flv -f avi -y md5:

Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with the MD5 output protocol.

UNIX pipe access protocol.

Read and write from UNIX pipes.

The accepted syntax is:

pipe:[<number>]

number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not specified, by default the stdout file descriptor will be used for writing, stdin for reading.

For example to read from stdin with ffmpeg:

cat test.wav | ffmpeg -i pipe:0 # …this is the same as… cat test.wav

For writing to stdout with ffmpeg:

ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi # …this is the same as… ffmpeg -i test.wav -f avi pipe: | cat > test.avi

This protocol accepts the following options:

blocksize

Set I/O operation maximum block size, in bytes. Default value is INT_MAX, which results in not limiting the requested block size. Setting this value reasonably low improves user termination request reaction time, which is valuable if data transmission is slow.

Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol.

Pro-MPEG Code of Practice #3 Release 2 FEC protocol.

The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism for MPEG-2 Transport Streams sent over RTP.

This protocol must be used in conjunction with the rtp_mpegts muxer and the rtp protocol.

The required syntax is:

-f rtp_mpegts -fec prompeg=<option>=<val>… rtp://<hostname>:<port>

The destination UDP ports are port + 2 for the column FEC stream and port + 4 for the row FEC stream.

This protocol accepts the following options:

l=n

The number of columns (4-20, LxD <= 100)

d=n

The number of rows (4-20, LxD <= 100)

Example usage:

-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>

Reliable Internet Streaming Transport protocol

The accepted options are:

rist_profile

Supported values:

simple

main

This one is default.

advanced

buffer_size

Set internal RIST buffer size in milliseconds for retransmission of data. Default value is 0 which means the librist default (1 sec). Maximum value is 30 seconds.

pkt_size

Set maximum packet size for sending data. 1316 by default.

log_level

Set loglevel for RIST logging messages. You only need to set this if you explicitly want to enable debug level messages or packet loss simulation, otherwise the regular loglevel is respected.

secret

Set override of encryption secret, by default is unset.

encryption

Set encryption type, by default is disabled. Acceptable values are 128 and 256.

Real-Time Messaging Protocol.

The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a TCP/IP network.

The required syntax is:

rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]

The accepted parameters are:

username

An optional username (mostly for publishing).

password

An optional password (mostly for publishing).

server

The address of the RTMP server.

port

The number of the TCP port to use (by default is 1935).

app

It is the name of the application to access. It usually corresponds to the path where the application is installed on the RTMP server (e.g. ondemand, flash/live, etc.). You can override the value parsed from the URI through the rtmp_app option, too.

playpath

It is the path or name of the resource to play with reference to the application specified in app, may be prefixed by mp4:. You can override the value parsed from the URI through the rtmp_playpath option, too.

listen

Act as a server, listening for an incoming connection.

timeout

Maximum time to wait for the incoming connection. Implies listen.

Additionally, the following parameters can be set via command line options (or in code via =AVOption=s):

rtmp_app

Name of application to connect on the RTMP server. This option overrides the parameter specified in the URI.

rtmp_buffer

Set the client buffer time in milliseconds. The default is 3000.

rtmp_conn

Extra arbitrary AMF connection parameters, parsed from a string, e.g. like B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0. Each value is prefixed by a single character denoting the type, B for Boolean, N for number, S for string, O for object, or Z for null, followed by a colon. For Booleans the data must be either 0 or 1 for FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or 1 to end or begin an object, respectively. Data items in subobjects may be named, by prefixing the type with ’N’ and specifying the name before the value (i.e. NB:myFlag:1). This option may be used multiple times to construct arbitrary AMF sequences.

rtmp_flashver

Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)

rtmp_flush_interval

Number of packets flushed in the same request (RTMPT only). The default is 10.

rtmp_live

Specify that the media is a live stream. No resuming or seeking in live streams is possible. The default value is any, which means the subscriber first tries to play the live stream specified in the playpath. If a live stream of that name is not found, it plays the recorded stream. The other possible values are live and recorded.

rtmp_pageurl

URL of the web page in which the media was embedded. By default no value will be sent.

rtmp_playpath

Stream identifier to play or to publish. This option overrides the parameter specified in the URI.

rtmp_subscribe

Name of live stream to subscribe to. By default no value will be sent. It is only sent if the option is specified or if rtmp_live is set to live.

rtmp_swfhash

SHA256 hash of the decompressed SWF file (32 bytes).

rtmp_swfsize

Size of the decompressed SWF file, required for SWFVerification.

rtmp_swfurl

URL of the SWF player for the media. By default no value will be sent.

rtmp_swfverify

URL to player swf file, compute hash/size automatically.

rtmp_tcurl

URL of the target stream. Defaults to proto://host[:port]/app.

For example to read with ffplay a multimedia resource named sample from the application vod from an RTMP server myserver:

ffplay rtmp://myserver/vod/sample

To publish to a password protected server, passing the playpath and app names separately:

ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

Encrypted Real-Time Messaging Protocol.

The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia content within standard cryptographic primitives, consisting of Diffie-Hellman key exchange and HMACSHA256, generating a pair of RC4 keys.

Real-Time Messaging Protocol over a secure SSL connection.

The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across an encrypted connection.

Real-Time Messaging Protocol tunneled through HTTP.

The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming multimedia content within HTTP requests to traverse firewalls.

Encrypted Real-Time Messaging Protocol tunneled through HTTP.

The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for streaming multimedia content within HTTP requests to traverse firewalls.

Real-Time Messaging Protocol tunneled through HTTPS.

The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming multimedia content within HTTPS requests to traverse firewalls.

libsmbclient permits one to manipulate CIFS/SMB network resources.

Following syntax is required.

smb://[[domain:]user[:password@]]server[/share[/path[/file]]]

This protocol accepts the following options.

timeout

Set timeout in milliseconds of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified.

truncate

Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.

workgroup

Set the workgroup used for making connections. By default workgroup is not specified.

For more information see: <*http://www.samba.org/*>.

Secure File Transfer Protocol via libssh

Read from or write to remote resources using SFTP protocol.

Following syntax is required.

sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

This protocol accepts the following options.

timeout

Set timeout of socket I/O operations used by the underlying low level operation. By default it is set to -1, which means that the timeout is not specified.

truncate

Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.

private_key

Specify the path of the file containing private key to use during authorization. By default libssh searches for keys in the ~.ssh// directory.

Example: Play a file stored on remote server.

ffplay sftp://user:password@server_address:22/home/user/resource.mpeg

Real-Time Messaging Protocol and its variants supported through librtmp.

Requires the presence of the librtmp headers and library during configuration. You need to explicitly configure the build with –enable-librtmp. If enabled this will replace the native RTMP protocol.

This protocol provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS).

The required syntax is:

<rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>

where rtmp_proto is one of the strings rtmp, rtmpt, rtmpe, rtmps, rtmpte, rtmpts corresponding to each RTMP variant, and server, port, app and playpath have the same meaning as specified for the RTMP native protocol. options contains a list of space-separated options of the form key/=/val.

See the librtmp manual page (man 3 librtmp) for more information.

For example, to stream a file in real-time to an RTMP server using ffmpeg:

ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream

To play the same stream using ffplay:

ffplay “rtmp://myserver/live/mystream live=1”

Real-time Transport Protocol.

The required syntax for an RTP URL is: rtp:///hostname/[:/port/][?/option/=/val/…]

port specifies the RTP port to use.

The following URL options are supported:

ttl=n

Set the TTL (Time-To-Live) value (for multicast only).

rtcpport=n

Set the remote RTCP port to n.

localrtpport=n

Set the local RTP port to n.

localrtcpport=n’

Set the local RTCP port to n.

pkt_size=n

Set max packet size (in bytes) to n.

buffer_size=size

Set the maximum UDP socket buffer size in bytes.

connect=0|1

Do a connect() on the UDP socket (if set to 1) or not (if set to 0).

sources=ip[,ip]

List allowed source IP addresses.

block=ip[,ip]

List disallowed (blocked) source IP addresses.

write_to_source=0|1

Send packets to the source address of the latest received packet (if set to 1) or to a default remote address (if set to 0).

localport=n

Set the local RTP port to n.

timeout=n

Set timeout (in microseconds) of socket I/O operations to n. This is a deprecated option. Instead, localrtpport should be used.

Important notes:

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

Real-Time Streaming Protocol.

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

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

The required syntax for a RTSP url is:

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

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

The following options are supported.

initial_pause

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

rtsp_transport

Set RTSP transport protocols. It accepts the following values:

udp

Use UDP as lower transport protocol.

tcp

Use TCP (interleaving within the RTSP control channel) as lower transport protocol.

udp_multicast

Use UDP multicast as lower transport protocol.

http

Use HTTP tunneling as lower transport protocol, which is useful for passing proxies.

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

rtsp_flags

Set RTSP flags. The following values are accepted:

filter_src

Accept packets only from negotiated peer address and port.

listen

Act as a server, listening for an incoming connection.

prefer_tcp

Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.

Default value is none.

allowed_media_types

Set media types to accept from the server. The following flags are accepted:

video

audio

data

By default it accepts all media types.

min_port

Set minimum local UDP port. Default value is 5000.

max_port

Set maximum local UDP port. Default value is 65000.

timeout

Set maximum timeout (in seconds) to wait for incoming connections. A value of -1 means infinite (default). This option implies the rtsp_flags set to listen.

reorder_queue_size

Set number of packets to buffer for handling of reordered packets.

stimeout

Set socket TCP I/O timeout in microseconds.

user-agent

Override User-Agent header. If not specified, it defaults to the libavformat identifier string.

When receiving data over UDP, the demuxer tries to reorder received packets (since they may arrive out of order, or packets may get lost totally). This can be disabled by setting the maximum demuxing delay to zero (via the max_delay field of AVFormatContext).

When watching multi-bitrate Real-RTSP streams with ffplay, the streams to display can be chosen with -vst n and -ast n for video and audio respectively, and can be switched on the fly by pressing v and a.

Examples

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

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

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

Muxer

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

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

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

announce_addr=address

Specify the destination IP address for sending the announcements to. If omitted, the announcements are sent to the commonly used SAP announcement multicast address 224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.

announce_port=port

Specify the port to send the announcements on, defaults to 9875 if not specified.

ttl=ttl

Specify the time to live value for the announcements and RTP packets, defaults to 255.

same_port=0|1

If set to 1, send all RTP streams on the same port pair. If zero (the default), all streams are sent on unique ports, with each stream on a port 2 numbers higher than the previous. VLC/Live555 requires this to be set to 1, to be able to receive the stream. The RTP stack in libavformat for receiving requires all streams to be sent on unique ports.

Example command lines follow.

To broadcast a stream on the local subnet, for watching in VLC:

ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1

Similarly, for watching in ffplay:

ffmpeg -re -i <input> -f sap sap://224.0.0.255

And for watching in ffplay, over IPv6:

ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]

Demuxer

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

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

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

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

Example command lines follow.

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

ffplay sap://

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

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

Stream Control Transmission Protocol.

The accepted URL syntax is:

sctp://<host>:<port>[?<options>]

The protocol accepts the following options:

listen

If set to any value, listen for an incoming connection. Outgoing connection is done by default.

max_streams

Set the maximum number of streams. By default no limit is set.

Haivision Secure Reliable Transport Protocol via libsrt.

The supported syntax for a SRT URL is:

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

options contains a list of &-separated options of the form key/=/val.

or

<options> srt://<hostname>:<port>

options contains a list of ’-key val’ options.

This protocol accepts the following options.

connect_timeout=milliseconds

Connection timeout; SRT cannot connect for RTT > 1500 msec (2 handshake exchanges) with the default connect timeout of 3 seconds. This option applies to the caller and rendezvous connection modes. The connect timeout is 10 times the value set for the rendezvous mode (which can be used as a workaround for this connection problem with earlier versions).

ffs=bytes

Flight Flag Size (Window Size), in bytes. FFS is actually an internal parameter and you should set it to not less than recv_buffer_size and mss. The default value is relatively large, therefore unless you set a very large receiver buffer, you do not need to change this option. Default value is 25600.

inputbw=bytes/seconds

Sender nominal input rate, in bytes per seconds. Used along with oheadbw, when maxbw is set to relative (0), to calculate maximum sending rate when recovery packets are sent along with the main media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set while maxbw is set to relative (0), the actual input rate is evaluated inside the library. Default value is 0.

iptos=tos

IP Type of Service. Applies to sender only. Default value is 0xB8.

ipttl=ttl

IP Time To Live. Applies to sender only. Default value is 64.

latency=microseconds

Timestamp-based Packet Delivery Delay. Used to absorb bursts of missed packet retransmissions. This flag sets both rcvlatency and peerlatency to the same value. Note that prior to version 1.3.0 this is the only flag to set the latency, however this is effectively equivalent to setting peerlatency, when side is sender and rcvlatency when side is receiver, and the bidirectional stream sending is not supported.

listen_timeout=microseconds

Set socket listen timeout.

maxbw=bytes/seconds

Maximum sending bandwidth, in bytes per seconds. -1 infinite (CSRTCC limit is 30mbps) 0 relative to input rate (see inputbw) >0 absolute limit value Default value is 0 (relative)

mode=caller|listener|rendezvous

Connection mode. caller opens client connection. listener starts server to listen for incoming connections. rendezvous use Rendez-Vous connection mode. Default value is caller.

mss=bytes

Maximum Segment Size, in bytes. Used for buffer allocation and rate calculation using a packet counter assuming fully filled packets. The smallest MSS between the peers is used. This is 1500 by default in the overall internet. This is the maximum size of the UDP packet and can be only decreased, unless you have some unusual dedicated network settings. Default value is 1500.

nakreport=1|0

If set to 1, Receiver will send `UMSG_LOSSREPORT` messages periodically until a lost packet is retransmitted or intentionally dropped. Default value is 1.

oheadbw=percents

Recovery bandwidth overhead above input rate, in percents. See inputbw. Default value is 25%.

passphrase=string

HaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. The passphrase is the shared secret between the sender and the receiver. It is used to generate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function). It is used only if pbkeylen is non-zero. It is used on the receiver only if the received data is encrypted. The configured passphrase cannot be recovered (write-only).

enforced_encryption=1|0

If true, both connection parties must have the same password set (including empty, that is, with no encryption). If the password doesn’t match or only one side is unencrypted, the connection is rejected. Default is true.

kmrefreshrate=packets

The number of packets to be transmitted after which the encryption key is switched to a new key. Default is -1. -1 means auto (0x1000000 in srt library). The range for this option is integers in the 0 - INT_MAX.

kmpreannounce=packets

The interval between when a new encryption key is sent and when switchover occurs. This value also applies to the subsequent interval between when switchover occurs and when the old encryption key is decommissioned. Default is -1. -1 means auto (0x1000 in srt library). The range for this option is integers in the 0 - INT_MAX.

payload_size=bytes

Sets the maximum declared size of a packet transferred during the single call to the sending function in Live mode. Use 0 if this value isn’t used (which is default in file mode). Default is -1 (automatic), which typically means MPEG-TS; if you are going to use SRT to send any different kind of payload, such as, for example, wrapping a live stream in very small frames, then you can use a bigger maximum frame size, though not greater than 1456 bytes.

pkt_size=bytes

Alias for payload_size.

peerlatency=microseconds

The latency value (as described in rcvlatency) that is set by the sender side as a minimum value for the receiver.

pbkeylen=bytes

Sender encryption key length, in bytes. Only can be set to 0, 16, 24 and 32. Enable sender encryption if not 0. Not required on receiver (set to 0), key size obtained from sender in HaiCrypt handshake. Default value is 0.

rcvlatency=microseconds

The time that should elapse since the moment when the packet was sent and the moment when it’s delivered to the receiver application in the receiving function. This time should be a buffer time large enough to cover the time spent for sending, unexpectedly extended RTT time, and the time needed to retransmit the lost UDP packet. The effective latency value will be the maximum of this options’ value and the value of peerlatency set by the peer side. Before version 1.3.0 this option is only available as latency.

recv_buffer_size=bytes

Set UDP receive buffer size, expressed in bytes.

send_buffer_size=bytes

Set UDP send buffer size, expressed in bytes.

timeout=microseconds

Set raise error timeouts for read, write and connect operations. Note that the SRT library has internal timeouts which can be controlled separately, the value set here is only a cap on those.

tlpktdrop=1|0

Too-late Packet Drop. When enabled on receiver, it skips missing packets that have not been delivered in time and delivers the following packets to the application when their time-to-play has come. It also sends a fake ACK to the sender. When enabled on sender and enabled on the receiving peer, the sender drops the older packets that have no chance of being delivered in time. It was automatically enabled in the sender if the receiver supports it.

sndbuf=bytes

Set send buffer size, expressed in bytes.

rcvbuf=bytes

Set receive buffer size, expressed in bytes. Receive buffer must not be greater than ffs.

lossmaxttl=packets

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

minversion

The minimum SRT version that is required from the peer. A connection to a peer that does not satisfy the minimum version requirement will be rejected. The version format in hex is 0xXXYYZZ for x.y.z in human readable form.

streamid=string

A string limited to 512 characters that can be set on the socket prior to connecting. This stream ID will be able to be retrieved by the listener side from the socket that is returned from srt_accept and was connected by a socket with that set stream ID. SRT does not enforce any special interpretation of the contents of this string. This option doesnXt make sense in Rendezvous connection; the result might be that simply one side will override the value from the other side and itXs the matter of luck which one would win

smoother=live|file

The type of Smoother used for the transmission for that socket, which is responsible for the transmission and congestion control. The Smoother type must be exactly the same on both connecting parties, otherwise the connection is rejected.

messageapi=1|0

When set, this socket uses the Message API, otherwise it uses Buffer API. Note that in live mode (see transtype) thereXs only message API available. In File mode you can chose to use one of two modes: Stream API (default, when this option is false). In this mode you may send as many data as you wish with one sending instruction, or even use dedicated functions that read directly from a file. The internal facility will take care of any speed and congestion control. When receiving, you can also receive as many data as desired, the data not extracted will be waiting for the next call. There is no boundary between data portions in the Stream mode. Message API. In this mode your single sending instruction passes exactly one piece of data that has boundaries (a message). Contrary to Live mode, this message may span across multiple UDP packets and the only size limitation is that it shall fit as a whole in the sending buffer. The receiver shall use as large buffer as necessary to receive the message, otherwise the message will not be given up. When the message is not complete (not all packets received or there was a packet loss) it will not be given up.

transtype=live|file

Sets the transmission type for the socket, in particular, setting this option sets multiple other parameters to their default values as required for a particular transmission type. live: Set options as for live transmission. In this mode, you should send by one sending instruction only so many data that fit in one UDP packet, and limited to the value defined first in payload_size (1316 is default in this mode). There is no speed control in this mode, only the bandwidth control, if configured, in order to not exceed the bandwidth with the overhead transmission (retransmitted and control packets). file: Set options as for non-live transmission. See messageapi for further explanations

linger=seconds

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

For more information see: <*https://github.com/Haivision/srt*>.

Secure Real-time Transport Protocol.

The accepted options are:

srtp_in_suite

srtp_out_suite

Select input and output encoding suites. Supported values:

AES_CM_128_HMAC_SHA1_80

SRTP_AES128_CM_HMAC_SHA1_80

AES_CM_128_HMAC_SHA1_32

SRTP_AES128_CM_HMAC_SHA1_32

nil

srtp_in_params

srtp_out_params

Set input and output encoding parameters, which are expressed by a base64-encoded representation of a binary block. The first 16 bytes of this binary block are used as master key, the following 14 bytes are used as master salt.

Virtually extract a segment of a file or another stream. The underlying stream must be seekable.

Accepted options:

start

Start offset of the extracted segment, in bytes.

end

End offset of the extracted segment, in bytes. If set to 0, extract till end of file.

Examples:

Extract a chapter from a DVD VOB file (start and end sectors obtained externally and multiplied by 2048):

subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB

Play an AVI file directly from a TAR archive:

subfile,,start,183241728,end,366490624,,:archive.tar

Play a MPEG-TS file from start offset till end:

subfile,,start,32815239,end,0,,:video.ts

Writes the output to multiple protocols. The individual outputs are separated by |

tee:file://path/to/local/this.avi|file://path/to/local/that.avi

Transmission Control Protocol.

The required syntax for a TCP url is:

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

options contains a list of &-separated options of the form key/=/val.

The list of supported options follows.

listen=2|1|0

Listen for an incoming connection. 0 disables listen, 1 enables listen in single client mode, 2 enables listen in multi-client mode. Default value is 0.

timeout=microseconds

Set raise error timeout, expressed in microseconds. This option is only relevant in read mode: if no data arrived in more than this time interval, raise error.

listen_timeout=milliseconds

Set listen timeout, expressed in milliseconds.

recv_buffer_size=bytes

Set receive buffer size, expressed bytes.

send_buffer_size=bytes

Set send buffer size, expressed bytes.

tcp_nodelay=1|0

Set TCP_NODELAY to disable Nagle’s algorithm. Default value is 0.

tcp_mss=bytes

Set maximum segment size for outgoing TCP packets, expressed in bytes.

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

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

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

The required syntax for a TLS/SSL url is:

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

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

ca_file, cafile=filename

A file containing certificate authority (CA) root certificates to treat as trusted. If the linked TLS library contains a default this might not need to be specified for verification to work, but not all libraries and setups have defaults built in. The file must be in OpenSSL PEM format.

tls_verify=1|0

If enabled, try to verify the peer that we are communicating with. Note, if using OpenSSL, this currently only makes sure that the peer certificate is signed by one of the root certificates in the CA database, but it does not validate that the certificate actually matches the host name we are trying to connect to. (With other backends, the host name is validated as well.) This is disabled by default since it requires a CA database to be provided by the caller in many cases.

cert_file, cert=filename

A file containing a certificate to use in the handshake with the peer. (When operating as server, in listen mode, this is more often required by the peer, while client certificates only are mandated in certain setups.)

key_file, key=filename

A file containing the private key for the certificate.

listen=1|0

If enabled, listen for connections on the provided port, and assume the server role in the handshake instead of the client role.

http_proxy

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

Example command lines:

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

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

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

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

User Datagram Protocol.

The required syntax for an UDP URL is:

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

options contains a list of &-separated options of the form key/=/val.

In case threading is enabled on the system, a circular buffer is used to store the incoming data, which allows one to reduce loss of data due to UDP socket buffer overruns. The fifo_size and overrun_nonfatal options are related to this buffer.

The list of supported options follows.

buffer_size=size

Set the UDP maximum socket buffer size in bytes. This is used to set either the receive or send buffer size, depending on what the socket is used for. Default is 32 KB for output, 384 KB for input. See also fifo_size.

bitrate=bitrate

If set to nonzero, the output will have the specified constant bitrate if the input has enough packets to sustain it.

burst_bits=bits

When using bitrate this specifies the maximum number of bits in packet bursts.

localport=port

Override the local UDP port to bind with.

localaddr=addr

Local IP address of a network interface used for sending packets or joining multicast groups.

pkt_size=size

Set the size in bytes of UDP packets.

reuse=1|0

Explicitly allow or disallow reusing UDP sockets.

ttl=ttl

Set the time to live value (for multicast only).

connect=1|0

Initialize the UDP socket with connect(). In this case, the destination address can’t be changed with ff_udp_set_remote_url later. If the destination address isn’t known at the start, this option can be specified in ff_udp_set_remote_url, too. This allows finding out the source address for the packets with getsockname, and makes writes return with AVERROR(ECONNREFUSED) if destination unreachable is received. For receiving, this gives the benefit of only receiving packets from the specified peer address/port.

sources=address[,address]

Only receive packets sent from the specified addresses. In case of multicast, also subscribe to multicast traffic coming from these addresses only.

block=address[,address]

Ignore packets sent from the specified addresses. In case of multicast, also exclude the source addresses in the multicast subscription.

fifo_size=units

Set the UDP receiving circular buffer size, expressed as a number of packets with size of 188 bytes. If not specified defaults to 7*4096.

overrun_nonfatal=1|0

Survive in case of UDP receiving circular buffer overrun. Default value is 0.

timeout=microseconds

Set raise error timeout, expressed in microseconds. This option is only relevant in read mode: if no data arrived in more than this time interval, raise error.

broadcast=1|0

Explicitly allow or disallow UDP broadcasting. Note that broadcasting may not work properly on networks having a broadcast storm protection.

Examples

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

Unix local socket

The required syntax for a Unix socket URL is:

unix://<filepath>

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

timeout

Timeout in ms.

listen

Create the Unix socket in listening mode.

ZeroMQ asynchronous messaging using the libzmq library.

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

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

zmq:tcp://ip-address:port

Example: Create a localhost stream on port 5555:

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

Multiple clients may connect to the stream using:

ffplay zmq:tcp://127.0.0.1:5555

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

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

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

pkt_size

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

ALSA (Advanced Linux Sound Architecture) input device.

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

This device allows capturing from an ALSA device. The name of the device to capture has to be an ALSA card identifier.

An ALSA identifier has the syntax:

hw:<CARD>[,<DEV>[,<SUBDEV>]]

where the DEV and SUBDEV components are optional.

The three arguments (in order: CARD,/DEV/,/SUBDEV/) specify card number or identifier, device number and subdevice number (-1 means any).

To see the list of cards currently recognized by your system check the files /proc/asound/cards and /proc/asound/devices.

For example to capture with ffmpeg from an ALSA device with card id 0, you may run the command:

ffmpeg -f alsa -i hw:0 alsaout.wav

For more information see: <*http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html*>

Options

sample_rate

Set the sample rate in Hz. Default is 48000.

channels

Set the number of channels. Default is 2.

Android camera input device.

This input devices uses the Android Camera2 NDK API which is available on devices with API level 24+. The availability of android_camera is autodetected during configuration.

This device allows capturing from all cameras on an Android device, which are integrated into the Camera2 NDK API.

The available cameras are enumerated internally and can be selected with the camera_index parameter. The input file string is discarded.

Generally the back facing camera has index 0 while the front facing camera has index 1.

Options

video_size

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

framerate

Set the video framerate. Falls back to the first available configuration reported by Android if requested framerate is not available or by default (-1).

camera_index

Set the index of the camera to use. Default is 0.

input_queue_size

Set the maximum number of frames to buffer. Default is 5.

AVFoundation input device.

AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.

The input filename has to be given in the following syntax:

-i “[[VIDEO]:[AUDIO]]”

The first entry selects the video input while the latter selects the audio input. The stream has to be specified by the device name or the device index as shown by the device list. Alternatively, the video and/or audio input device can be chosen by index using the

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

and/or

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

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

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

There are two device name aliases:

“default”

Select the AVFoundation default device of the corresponding type.

“none”

Do not record the corresponding media type. This is equivalent to specifying an empty device name or index.

Options

AVFoundation supports the following options:

-list_devices <TRUE|FALSE>

If set to true, a list of all available input devices is given showing all device names and indices.

-video_device_index <INDEX>

Specify the video device by its index. Overrides anything given in the input filename.

-audio_device_index <INDEX>

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

-pixel_format <FORMAT>

Request the video device to use a specific pixel format. If the specified format is not supported, a list of available formats is given and the first one in this list is used instead. Available pixel formats are: monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0, bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10, yuv420p, nv12, yuyv422, gray

-framerate

Set the grabbing frame rate. Default is ntsc, corresponding to a frame rate of 30000/1001.

-video_size

Set the video frame size.

-capture_cursor

Capture the mouse pointer. Default is 0.

-capture_mouse_clicks

Capture the screen mouse clicks. Default is 0.

-capture_raw_data

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

Examples

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

BSD video input device.

Options

framerate

Set the frame rate.

video_size

Set the video frame size. Default is vga.

standard

Available values are:

pal

ntsc

secam

paln

palm

ntscj

The decklink input device provides capture capabilities for Blackmagic DeckLink devices.

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

DeckLink is very picky about the formats it supports. Pixel format of the input can be set with raw_format. Framerate and video size must be determined for your device with -list_formats 1. Audio sample rate is always 48 kHz and the number of channels can be 2, 8 or 16. Note that all audio channels are bundled in one single audio track.

Options

list_devices

If set to true, print a list of devices and exit. Defaults to false. This option is deprecated, please use the -sources option of ffmpeg to list the available input devices.

list_formats

If set to true, print a list of supported formats and exit. Defaults to false.

format_code <FourCC>

This sets the input video format to the format given by the FourCC. To see the supported values of your device(s) use list_formats. Note that there is a FourCC ’pal ’ that can also be used as pal (3 letters). Default behavior is autodetection of the input video format, if the hardware supports it.

raw_format

Set the pixel format of the captured video. Available values are:

auto

This is the default which means 8-bit YUV 422 or 8-bit ARGB if format autodetection is used, 8-bit YUV 422 otherwise.

uyvy422

8-bit YUV 422.

yuv422p10

10-bit YUV 422.

argb

8-bit RGB.

bgra

8-bit RGB.

rgb10

10-bit RGB.

teletext_lines

If set to nonzero, an additional teletext stream will be captured from the vertical ancillary data. Both SD PAL (576i) and HD (1080i or 1080p) sources are supported. In case of HD sources, OP47 packets are decoded. This option is a bitmask of the SD PAL VBI lines captured, specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB in the mask. Selected lines which do not contain teletext information will be ignored. You can use the special all constant to select all possible lines, or standard to skip lines 6, 318 and 319, which are not compatible with all receivers. For SD sources, ffmpeg needs to be compiled with --enable-libzvbi. For HD sources, on older (pre-4K) DeckLink card models you have to capture in 10 bit mode.

channels

Defines number of audio channels to capture. Must be 2, 8 or 16. Defaults to 2.

duplex_mode

Sets the decklink device duplex mode. Must be unset, half or full. Defaults to unset.

timecode_format

Timecode type to include in the frame and video stream metadata. Must be none, rp188vitc, rp188vitc2, rp188ltc, rp188hfr, rp188any, vitc, vitc2, or serial. Defaults to none (not included). In order to properly support 50/60 fps timecodes, the ordering of the queried timecode types for rp188any is HFR, VITC1, VITC2 and LTC for >30 fps content. Note that this is slightly different to the ordering used by the DeckLink API, which is HFR, VITC1, LTC, VITC2.

video_input

Sets the video input source. Must be unset, sdi, hdmi, optical_sdi, component, composite or s_video. Defaults to unset.

audio_input

Sets the audio input source. Must be unset, embedded, aes_ebu, analog, analog_xlr, analog_rca or microphone. Defaults to unset.

video_pts

Sets the video packet timestamp source. Must be video, audio, reference, wallclock or abs_wallclock. Defaults to video.

audio_pts

Sets the audio packet timestamp source. Must be video, audio, reference, wallclock or abs_wallclock. Defaults to audio.

draw_bars

If set to true, color bars are drawn in the event of a signal loss. Defaults to true.

queue_size

Sets maximum input buffer size in bytes. If the buffering reaches this value, incoming frames will be dropped. Defaults to 1073741824.

audio_depth

Sets the audio sample bit depth. Must be 16 or 32. Defaults to 16.

decklink_copyts

If set to true, timestamps are forwarded as they are without removing the initial offset. Defaults to false.

timestamp_align

Capture start time alignment in seconds. If set to nonzero, input frames are dropped till the system timestamp aligns with configured value. Alignment difference of up to one frame duration is tolerated. This is useful for maintaining input synchronization across N different hardware devices deployed for ’N-way’ redundancy. The system time of different hardware devices should be synchronized with protocols such as NTP or PTP, before using this option. Note that this method is not foolproof. In some border cases input synchronization may not happen due to thread scheduling jitters in the OS. Either sync could go wrong by 1 frame or in a rarer case timestamp_align seconds. Defaults to 0.

wait_for_tc (bool)

Drop frames till a frame with timecode is received. Sometimes serial timecode isn’t received with the first input frame. If that happens, the stored stream timecode will be inaccurate. If this option is set to true, input frames are dropped till a frame with timecode is received. Option timecode_format must be specified. Defaults to false.

enable_klv(bool)

If set to true, extracts KLV data from VANC and outputs KLV packets. KLV VANC packets are joined based on MID and PSC fields and aggregated into one KLV packet. Defaults to false.

Examples

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

Windows DirectShow input device.

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

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

The input name should be in the format:

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

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

Options

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

video_size

Set the video size in the captured video.

framerate

Set the frame rate in the captured video.

sample_rate

Set the sample rate (in Hz) of the captured audio.

sample_size

Set the sample size (in bits) of the captured audio.

channels

Set the number of channels in the captured audio.

list_devices

If set to true, print a list of devices and exit.

list_options

If set to true, print a list of selected device’s options and exit.

video_device_number

Set video device number for devices with the same name (starts at 0, defaults to 0).

audio_device_number

Set audio device number for devices with the same name (starts at 0, defaults to 0).

pixel_format

Select pixel format to be used by DirectShow. This may only be set when the video codec is not set or set to rawvideo.

audio_buffer_size

Set audio device buffer size in milliseconds (which can directly impact latency, depending on the device). Defaults to using the audio device’s default buffer size (typically some multiple of 500ms). Setting this value too low can degrade performance. See also <*http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx*>

video_pin_name

Select video capture pin to use by name or alternative name.

audio_pin_name

Select audio capture pin to use by name or alternative name.

crossbar_video_input_pin_number

Select video input pin number for crossbar device. This will be routed to the crossbar device’s Video Decoder output pin. Note that changing this value can affect future invocations (sets a new default) until system reboot occurs.

crossbar_audio_input_pin_number

Select audio input pin number for crossbar device. This will be routed to the crossbar device’s Audio Decoder output pin. Note that changing this value can affect future invocations (sets a new default) until system reboot occurs.

show_video_device_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to change video filter properties and configurations manually. Note that for crossbar devices, adjusting values in this dialog may be needed at times to toggle between PAL (25 fps) and NTSC (29.97) input frame rates, sizes, interlacing, etc. Changing these values can enable different scan rates/frame rates and avoiding green bars at the bottom, flickering scan lines, etc. Note that with some devices, changing these properties can also affect future invocations (sets new defaults) until system reboot occurs.

show_audio_device_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to change audio filter properties and configurations manually.

show_video_crossbar_connection_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify crossbar pin routings, when it opens a video device.

show_audio_crossbar_connection_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify crossbar pin routings, when it opens an audio device.

show_analog_tv_tuner_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify TV channels and frequencies.

show_analog_tv_tuner_audio_dialog

If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify TV audio (like mono vs. stereo, Language A,B or C).

audio_device_load

Load an audio capture filter device from file instead of searching it by name. It may load additional parameters too, if the filter supports the serialization of its properties to. To use this an audio capture source has to be specified, but it can be anything even fake one.

audio_device_save

Save the currently used audio capture filter device and its parameters (if the filter supports it) to a file. If a file with the same name exists it will be overwritten.

video_device_load

Load a video capture filter device from file instead of searching it by name. It may load additional parameters too, if the filter supports the serialization of its properties to. To use this a video capture source has to be specified, but it can be anything even fake one.

video_device_save

Save the currently used video capture filter device and its parameters (if the filter supports it) to a file. If a file with the same name exists it will be overwritten.

Examples

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

Linux framebuffer input device.

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

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

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

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

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

You can take a single screenshot image with the command:

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

Options

framerate

Set the frame rate. Default is 25.

Win32 GDI-based screen capture device.

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

There are two options for the input filename:

desktop

or

title=<window_title>

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

For example, to grab the entire desktop using ffmpeg:

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

Grab a 640x480 region at position 10,20:

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

Grab the contents of the window named Calculator

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

Options

draw_mouse

Specify whether to draw the mouse pointer. Use the value 0 to not draw the pointer. Default value is 1.

framerate

Set the grabbing frame rate. Default value is ntsc, corresponding to a frame rate of 30000/1001.

show_region

Show grabbed region on screen. If show_region is specified with 1, then the grabbing region will be indicated on screen. With this option, it is easy to know what is being grabbed if only a portion of the screen is grabbed. Note that show_region is incompatible with grabbing the contents of a single window. For example: ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg

video_size

Set the video frame size. The default is to capture the full screen if desktop is selected, or the full window size if title=window_title is selected.

offset_x

When capturing a region with video_size, set the distance from the left edge of the screen or desktop. Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned to the left of your primary monitor, you will need to use a negative offset_x value to move the region to that monitor.

offset_y

When capturing a region with video_size, set the distance from the top edge of the screen or desktop. Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned above your primary monitor, you will need to use a negative offset_y value to move the region to that monitor.

FireWire DV/HDV input device using libiec61883.

To enable this input device, you need libiec61883, libraw1394 and libavc1394 installed on your system. Use the configure option --enable-libiec61883 to compile with the device enabled.

The iec61883 capture device supports capturing from a video device connected via IEEE1394 (FireWire), using libiec61883 and the new Linux FireWire stack (juju). This is the default DV/HDV input method in Linux Kernel 2.6.37 and later, since the old FireWire stack was removed.

Specify the FireWire port to be used as input file, or auto to choose the first port connected.

Options

dvtype

Override autodetection of DV/HDV. This should only be used if auto detection does not work, or if usage of a different device type should be prohibited. Treating a DV device as HDV (or vice versa) will not work and result in undefined behavior. The values auto, dv and hdv are supported.

dvbuffer

Set maximum size of buffer for incoming data, in frames. For DV, this is an exact value. For HDV, it is not frame exact, since HDV does not have a fixed frame size.

dvguid

Select the capture device by specifying its GUID. Capturing will only be performed from the specified device and fails if no device with the given GUID is found. This is useful to select the input if multiple devices are connected at the same time. Look at /sys/bus/firewire/devices to find out the GUIDs.

Examples

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

JACK input device.

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

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

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

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

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

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

ffmpeg -y out.wav # Start the sample jack_metro readable client. $ jack_metro -b 120 -d 0.2 -f 4000 # List the current JACK clients. $ jack_lsp -c system:capture_1 system:capture_2 system:playback_1 system:playback_2 ffmpeg:input_1 metro:120_bpm # Connect metro to the ffmpeg writable client. $ jack_connect metro:120_bpm ffmpeg:input_1

For more information read: <*http://jackaudio.org/*>

Options

channels

Set the number of channels. Default is 2.

KMS video input device.

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

Requires either DRM master or CAP_SYS_ADMIN to run.

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

Options

device

DRM device to capture on. Defaults to /dev/dri/card0.

format

Pixel format of the framebuffer. This can be autodetected if you are running Linux 5.7 or later, but needs to be provided for earlier versions. Defaults to bgr0, which is the most common format used by the Linux console and Xorg X server.

format_modifier

Format modifier to signal on output frames. This is necessary to import correctly into some APIs. It can be autodetected if you are running Linux 5.7 or later, but will need to be provided explicitly when needed in earlier versions. See the libdrm documentation for possible values.

crtc_id

KMS CRTC ID to define the capture source. The first active plane on the given CRTC will be used.

plane_id

KMS plane ID to define the capture source. Defaults to the first active plane found if neither crtc_id nor plane_id are specified.

framerate

Framerate to capture at. This is not synchronised to any page flipping or framebuffer changes - it just defines the interval at which the framebuffer is sampled. Sampling faster than the framebuffer update rate will generate independent frames with the same content. Defaults to 30.

Examples

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

Libavfilter input virtual device.

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

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

Options

graph

Specify the filtergraph to use as input. Each video open output must be labelled by a unique string of the form “out/N/”, where N is a number starting from 0 corresponding to the mapped input stream generated by the device. The first unlabelled output is automatically assigned to the out0 label, but all the others need to be specified explicitly. The suffix +subcc can be appended to the output label to create an extra stream with the closed captions packets attached to that output (experimental; only for EIA-608 / CEA-708 for now). The subcc streams are created after all the normal streams, in the order of the corresponding stream. For example, if there is out19+subcc, out7+subcc and up to out42, the stream #43 is subcc for stream #7 and stream #44 is subcc for stream #19. If not specified defaults to the filename specified for the input device.

graph_file

Set the filename of the filtergraph to be read and sent to the other filters. Syntax of the filtergraph is the same as the one specified by the option graph.

dumpgraph

Dump graph to stderr.

Examples

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

Audio-CD input device based on libcdio.

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

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

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

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

Options

speed

Set drive reading speed. Default value is 0. The speed is specified CD-ROM speed units. The speed is set through the libcdio cdio_cddap_speed_set function. On many CD-ROM drives, specifying a value too large will result in using the fastest speed.

paranoia_mode

Set paranoia recovery mode flags. It accepts one of the following values:

disable

verify

overlap

neverskip

full

Default value is disable. For more information about the available recovery modes, consult the paranoia project documentation.

IIDC1394 input device, based on libdc1394 and libraw1394.

Requires the configure option --enable-libdc1394.

Options

framerate

Set the frame rate. Default is ntsc, corresponding to a frame rate of 30000/1001.

pixel_format

Select the pixel format. Default is uyvy422.

video_size

Set the video size given as a string such as 640x480 or hd720. Default is qvga.

The OpenAL input device provides audio capture on all systems with a working OpenAL 1.1 implementation.

To enable this input device during configuration, you need OpenAL headers and libraries installed on your system, and need to configure FFmpeg with --enable-openal.

OpenAL headers and libraries should be provided as part of your OpenAL implementation, or as an additional download (an SDK). Depending on your installation you may need to specify additional flags via the --extra-cflags and --extra-ldflags for allowing the build system to locate the OpenAL headers and libraries.

An incomplete list of OpenAL implementations follows:

Creative

The official Windows implementation, providing hardware acceleration with supported devices and software fallback. See <*http://openal.org/*>.

OpenAL Soft

Portable, open source (LGPL) software implementation. Includes backends for the most common sound APIs on the Windows, Linux, Solaris, and BSD operating systems. See <*http://kcat.strangesoft.net/openal.html*>.

Apple

OpenAL is part of Core Audio, the official Mac OS X Audio interface. See <*http://developer.apple.com/technologies/mac/audio-and-video.html*>

This device allows one to capture from an audio input device handled through OpenAL.

You need to specify the name of the device to capture in the provided filename. If the empty string is provided, the device will automatically select the default device. You can get the list of the supported devices by using the option list_devices.

Options

channels

Set the number of channels in the captured audio. Only the values 1 (monaural) and 2 (stereo) are currently supported. Defaults to 2.

sample_size

Set the sample size (in bits) of the captured audio. Only the values 8 and 16 are currently supported. Defaults to 16.

sample_rate

Set the sample rate (in Hz) of the captured audio. Defaults to 44.1k.

list_devices

If set to true, print a list of devices and exit. Defaults to false.

Examples

Print the list of OpenAL supported devices and exit:

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

Capture from the OpenAL device DR-BT101 via PulseAudio:

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

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

$ ffmpeg -f openal -i out.ogg

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

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

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

Open Sound System input device.

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

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

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

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

Options

sample_rate

Set the sample rate in Hz. Default is 48000.

channels

Set the number of channels. Default is 2.

PulseAudio input device.

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

The filename to provide to the input device is a source device or the string default

To list the PulseAudio source devices and their properties you can invoke the command pactl list sources.

More information about PulseAudio can be found on <*http://www.pulseaudio.org*>.

Options

server

Connect to a specific PulseAudio server, specified by an IP address. Default server is used when not provided.

name

Specify the application name PulseAudio will use when showing active clients, by default it is the LIBAVFORMAT_IDENT string.

stream_name

Specify the stream name PulseAudio will use when showing active streams, by default it is record.

sample_rate

Specify the samplerate in Hz, by default 48kHz is used.

channels

Specify the channels in use, by default 2 (stereo) is set.

frame_size

Specify the number of bytes per frame, by default it is set to 1024.

fragment_size

Specify the minimal buffering fragment in PulseAudio, it will affect the audio latency. By default it is unset.

wallclock

Set the initial PTS using the current time. Default is 1.

Examples

Record a stream from default device:

ffmpeg -f pulse -i default /tmp/pulse.wav

sndio input device.

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

The filename to provide to the input device is the device node representing the sndio input device, and is usually set to /dev/audio0.

For example to grab from /dev/audio0 using ffmpeg use the command:

ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav

Options

sample_rate

Set the sample rate in Hz. Default is 48000.

channels

Set the number of channels. Default is 2.

Video4Linux2 input video device.

v4l2 can be used as alias for video4linux2.

If FFmpeg is built with v4l-utils support (by using the --enable-libv4l2 configure option), it is possible to use it with the -use_libv4l2 input device option.

The name of the device to grab is a file device node, usually Linux systems tend to automatically create such nodes when the device (e.g. an USB webcam) is plugged into the system, and has a name of the kind /dev/videoN, where N is a number associated to the device.

Video4Linux2 devices usually support a limited set of width/x/height sizes and frame rates. You can check which are supported using -list_formats all for Video4Linux2 devices. Some devices, like TV cards, support one or more standards. It is possible to list all the supported standards using -list_standards all.

The time base for the timestamps is 1 microsecond. Depending on the kernel version and configuration, the timestamps may be derived from the real time clock (origin at the Unix Epoch) or the monotonic clock (origin usually at boot time, unaffected by NTP or manual changes to the clock). The -timestamps abs or -ts abs option can be used to force conversion into the real time clock.

Some usage examples of the video4linux2 device with ffmpeg and ffplay:

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

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

Options

standard

Set the standard. Must be the name of a supported standard. To get a list of the supported standards, use the list_standards option.

channel

Set the input channel number. Default to -1, which means using the previously selected channel.

video_size

Set the video frame size. The argument must be a string in the form WIDTH/x/HEIGHT or a valid size abbreviation.

pixel_format

Select the pixel format (only valid for raw video input).

input_format

Set the preferred pixel format (for raw video) or a codec name. This option allows one to select the input format, when several are available.

framerate

Set the preferred video frame rate.

list_formats

List available formats (supported pixel formats, codecs, and frame sizes) and exit. Available values are:

all

Show all available (compressed and non-compressed) formats.

raw

Show only raw video (non-compressed) formats.

compressed

Show only compressed formats.

list_standards

List supported standards and exit. Available values are:

all

Show all supported standards.

timestamps, ts

Set type of timestamps for grabbed frames. Available values are:

default

Use timestamps from the kernel.

abs

Use absolute timestamps (wall clock).

mono2abs

Force conversion from monotonic to absolute timestamps.

Default value is default.

use_libv4l2

Use libv4l2 (v4l-utils) conversion functions. Default is 0.

VfW (Video for Windows) capture input device.

The filename passed as input is the capture driver number, ranging from 0 to 9. You may use list as filename to print a list of drivers. Any other filename will be interpreted as device number 0.

Options

video_size

Set the video frame size.

framerate

Set the grabbing frame rate. Default value is ntsc, corresponding to a frame rate of 30000/1001.

X11 video input device.

To enable this input device during configuration you need libxcb installed on your system. It will be automatically detected during configuration.

This device allows one to capture a region of an X11 display.

The filename passed as input has the syntax:

[<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]

hostname:/display_number/./screen_number/ specifies the X11 display name of the screen to grab from. hostname can be omitted, and defaults to localhost. The environment variable DISPLAY contains the default display name.

x_offset and y_offset specify the offsets of the grabbed area with respect to the top-left border of the X11 screen. They default to 0.

Check the X11 documentation (e.g. man X) for more detailed information.

Use the xdpyinfo program for getting basic information about the properties of your X11 display (e.g. grep for name or dimensions).

For example to grab from :0.0 using ffmpeg:

ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg

Grab at position 10,20:

ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

Options

select_region

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

draw_mouse

Specify whether to draw the mouse pointer. A value of 0 specifies not to draw the pointer. Default value is 1.

follow_mouse

Make the grabbed area follow the mouse. The argument can be centered or a number of pixels PIXELS. When it is specified with centered, the grabbing region follows the mouse pointer and keeps the pointer at the center of region; otherwise, the region follows only when the mouse pointer reaches within PIXELS (greater than zero) to the edge of region. For example: ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg To follow only when the mouse pointer reaches within 100 pixels to edge: ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg

framerate

Set the grabbing frame rate. Default value is ntsc, corresponding to a frame rate of 30000/1001.

show_region

Show grabbed region on screen. If show_region is specified with 1, then the grabbing region will be indicated on screen. With this option, it is easy to know what is being grabbed if only a portion of the screen is grabbed.

region_border

Set the region border thickness if -show_region 1 is used. Range is 1 to 128 and default is 3 (XCB-based x11grab only). For example: ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg With follow_mouse: ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg

window_id

Grab this window, instead of the whole screen. Default value is 0, which maps to the whole screen (root window). The id of a window can be found using the xwininfo program, possibly with options -tree and -root. If the window is later enlarged, the new area is not recorded. Video ends when the window is closed, unmapped (i.e., iconified) or shrunk beyond the video size (which defaults to the initial window size). This option disables options follow_mouse and select_region.

video_size

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

grab_x

grab_y

Set the grabbing region coordinates. They are expressed as offset from the top left corner of the X11 window and correspond to the x_offset and y_offset parameters in the device name. The default value for both options is 0.

A filtergraph has a textual representation, which is recognized by the -filter*/-vf*/*-af* and -filter_complex options in ffmpeg and -vf*/-af* in ffplay, and by the avfilter_graph_parse_ptr() function defined in libavfilter/avfilter.h.

A filterchain consists of a sequence of connected filters, each one connected to the previous one in the sequence. A filterchain is represented by a list of ,-separated filter descriptions.

A filtergraph consists of a sequence of filterchains. A sequence of filterchains is represented by a list of ;-separated filterchain descriptions.

A filter is represented by a string of the form: [/in_link_1/]…[/in_link_N/]/filter_name/@/id/=/arguments/[/out_link_1/]…[/out_link_M/]

filter_name is the name of the filter class of which the described filter is an instance of, and has to be the name of one of the filter classes registered in the program optionally followed by “@/id/. The name of the filter class is optionally followed by a string =/arguments/”.

arguments is a string which contains the parameters used to initialize the filter instance. It may have one of two forms:

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

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

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

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

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

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

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

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

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

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

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

Here is a BNF description of the filtergraph syntax:

<NAME> ::= sequence of alphanumeric characters and _ <FILTER_NAME> ::= <NAME>[“@”<NAME>] <LINKLABEL> ::= “[” <NAME> “]” <LINKLABELS> ::= <LINKLABEL> [<LINKLABELS>] <FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted) <FILTER> ::= [<LINKLABELS>] <FILTER_NAME> [“" <FILTER_ARGUMENTS>] [<LINKLABELS>] <FILTERCHAIN> :: <FILTER> [,<FILTERCHAIN>] <FILTERGRAPH> ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

Filtergraph description composition entails several levels of escaping. See the Quoting and escaping section in the ffmpeg-utils (1) manual for more information about the employed escaping procedure.

A first level escaping affects the content of each filter option value, which may contain the special character : used to separate values, or one of the escaping characters \.

A second level escaping affects the whole filter description, which may contain the escaping characters \ or the special characters [],; used by the filtergraph description.

Finally, when you specify a filtergraph on a shell commandline, you need to perform a third level escaping for the shell special characters contained within it.

For example, consider the following string to be embedded in the drawtext filter description text value:

this is a string: may contain one, or more, special characters

This string contains the special escaping character, and the : special character, so it needs to be escaped in this way:

text=this is a \string\\: may contain one, or more, special characters

A second level of escaping is required when embedding the filter description in a filtergraph description, in order to escape all the filtergraph special characters. Thus the example above becomes:

drawtext=text=this is a \\\string\\\\\: may contain one\, or more\, special characters

(note that in addition to the \ escaping special characters, also , needs to be escaped).

Finally an additional level of escaping is needed when writing the filtergraph description in a shell command, which depends on the escaping rules of the adopted shell. For example, assuming that \ is special and needs to be escaped with another \, the previous string will finally result in:

-vf “drawtext=text=this is a \\\\\\string\\\\\\\\\\: may contain one\\, or more\\, special characters”

A compressor is mainly used to reduce the dynamic range of a signal. Especially modern music is mostly compressed at a high ratio to improve the overall loudness. It’s done to get the highest attention of a listener, fatten the sound and bring more power to the track. If a signal is compressed too much it may sound dull or dead afterwards or it may start to pump (which could be a powerful effect but can also destroy a track completely). The right compression is the key to reach a professional sound and is the high art of mixing and mastering. Because of its complex settings it may take a long time to get the right feeling for this kind of effect.

Compression is done by detecting the volume above a chosen level threshold and dividing it by the factor set with ratio. So if you set the threshold to -12dB and your signal reaches -6dB a ratio of 2:1 will result in a signal at -9dB. Because an exact manipulation of the signal would cause distortion of the waveform the reduction can be levelled over the time. This is done by setting Attack and Release. attack determines how long the signal has to rise above the threshold before any reduction will occur and release sets the time the signal has to fall below the threshold to reduce the reduction again. Shorter signals than the chosen attack time will be left untouched. The overall reduction of the signal can be made up afterwards with the makeup setting. So compressing the peaks of a signal about 6dB and raising the makeup to this level results in a signal twice as loud than the source. To gain a softer entry in the compression the knee flattens the hard edge at the threshold in the range of the chosen decibels.

The filter accepts the following options:

level_in

Set input gain. Default is 1. Range is between 0.015625 and 64.

mode

Set mode of compressor operation. Can be upward or downward. Default is downward.

threshold

If a signal of stream rises above this level it will affect the gain reduction. By default it is 0.125. Range is between 0.00097563 and 1.

ratio

Set a ratio by which the signal is reduced. 1:2 means that if the level rose 4dB above the threshold, it will be only 2dB above after the reduction. Default is 2. Range is between 1 and 20.

attack

Amount of milliseconds the signal has to rise above the threshold before gain reduction starts. Default is 20. Range is between 0.01 and 2000.

release

Amount of milliseconds the signal has to fall below the threshold before reduction is decreased again. Default is 250. Range is between 0.01 and 9000.

makeup

Set the amount by how much signal will be amplified after processing. Default is 1. Range is from 1 to 64.

knee

Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.

link

Choose if the average level between all channels of input stream or the louder(maximum) channel of input stream affects the reduction. Default is average.

detection

Should the exact signal be taken in case of peak or an RMS one in case of rms. Default is rms which is mostly smoother.

(no term)

How much to use compressed signal in output. Default is 1. Range is between 0 and 1.

Commands

This filter supports the all above options as commands.

Simple audio dynamic range compression/expansion filter.

The filter accepts the following options:

contrast

Set contrast. Default is 33. Allowed range is between 0 and 100.

Copy the input audio source unchanged to the output. This is mainly useful for testing purposes.

Apply cross fade from one input audio stream to another input audio stream. The cross fade is applied for specified duration near the end of first stream.

The filter accepts the following options:

nb_samples, ns

Specify the number of samples for which the cross fade effect has to last. At the end of the cross fade effect the first input audio will be completely silent. Default is 44100.

duration, d

Specify the duration of the cross fade effect. See the Time duration section in the ffmpeg-utils (1) manual for the accepted syntax. By default the duration is determined by nb_samples. If set this option is used instead of nb_samples.

overlap, o

Should first stream end overlap with second stream start. Default is enabled.

curve1

Set curve for cross fade transition for first stream.

curve2

Set curve for cross fade transition for second stream. For description of available curve types see afade filter description.

Examples

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

Split audio stream into several bands.

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

The filter accepts the following options:

split

Set split frequencies. Those must be positive and increasing.

order

Set filter order for each band split. This controls filter roll-off or steepness of filter transfer function. Available values are:

2nd

12 dB per octave.

4th

24 dB per octave.

6th

36 dB per octave.

8th

48 dB per octave.

10th

60 dB per octave.

12th

72 dB per octave.

14th

84 dB per octave.

16th

96 dB per octave.

18th

108 dB per octave.

20th

120 dB per octave.

Default is 4th.

level

Set input gain level. Allowed range is from 0 to 1. Default value is 1.

gains

Set output gain for each band. Default value is 1 for all bands.

Examples

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

Reduce audio bit resolution.

This filter is bit crusher with enhanced functionality. A bit crusher is used to audibly reduce number of bits an audio signal is sampled with. This doesn’t change the bit depth at all, it just produces the effect. Material reduced in bit depth sounds more harsh and digital. This filter is able to even round to continuous values instead of discrete bit depths. Additionally it has a D/C offset which results in different crushing of the lower and the upper half of the signal. An Anti-Aliasing setting is able to produce softer crushing sounds.

Another feature of this filter is the logarithmic mode. This setting switches from linear distances between bits to logarithmic ones. The result is a much more natural sounding crusher which doesn’t gate low signals for example. The human ear has a logarithmic perception, so this kind of crushing is much more pleasant. Logarithmic crushing is also able to get anti-aliased.

The filter accepts the following options:

level_in

Set level in.

level_out

Set level out.

bits

Set bit reduction.

(no term)

Set mixing amount.

mode

Can be linear: lin or logarithmic: log.

(no term)

Set DC.

aa

Set anti-aliasing.

samples

Set sample reduction.

lfo

Enable LFO. By default disabled.

lforange

Set LFO range.

lforate

Set LFO rate.

Commands

This filter supports the all above options as commands.

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

Remove impulsive noise from input audio.

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

window, w

Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55 milliseconds. This sets size of window which will be processed at once.

overlap, o

Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default value is 75 percent. Setting this to a very high value increases impulsive noise removal but makes whole process much slower.

arorder, a

Set autoregression order, in percentage of window size. Allowed range is from 0 to 25. Default value is 2 percent. This option also controls quality of interpolated samples using neighbour good samples.

threshold, t

Set threshold value. Allowed range is from 1 to 100. Default value is 2. This controls the strength of impulsive noise which is going to be removed. The lower value, the more samples will be detected as impulsive noise.

burst, b

Set burst fusion, in percentage of window size. Allowed range is 0 to 10. Default value is 2. If any two samples detected as noise are spaced less than this value then any sample between those two samples will be also detected as noise.

method, m

Set overlap method. It accepts the following values:

add, a

Select overlap-add method. Even not interpolated samples are slightly changed with this method.

save, s

Select overlap-save method. Not interpolated samples remain unchanged.

Default value is a.

Remove clipped samples from input audio.

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

window, w

Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55 milliseconds. This sets size of window which will be processed at once.

overlap, o

Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default value is 75 percent.

arorder, a

Set autoregression order, in percentage of window size. Allowed range is from 0 to 25. Default value is 8 percent. This option also controls quality of interpolated samples using neighbour good samples.

threshold, t

Set threshold value. Allowed range is from 1 to 100. Default value is 10. Higher values make clip detection less aggressive.

hsize, n

Set size of histogram used to detect clips. Allowed range is from 100 to 9999. Default value is 1000. Higher values make clip detection less aggressive.

method, m

Set overlap method. It accepts the following values:

add, a

Select overlap-add method. Even not interpolated samples are slightly changed with this method.

save, s

Select overlap-save method. Not interpolated samples remain unchanged.

Default value is a.

Delay one or more audio channels.

Samples in delayed channel are filled with silence.

The filter accepts the following option:

delays

Set list of delays in milliseconds for each channel separated by ’|’. Unused delays will be silently ignored. If number of given delays is smaller than number of channels all remaining channels will not be delayed. If you want to delay exact number of samples, append ’S’ to number. If you want instead to delay in seconds, append ’s’ to number.

all

Use last set delay for all remaining channels. By default is disabled. This option if enabled changes how option delays is interpreted.

Examples

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

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

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

A description of the accepted parameters follows.

level

Set level of added noise in dB. Default is -351. Allowed range is from -451 to -90.

type

Set type of added noise.

1. Add DC signal.
2. Add AC signal.
3. Add square signal.
4. Add pulse signal.

Default is dc.

Commands

This filter supports the all above options as commands.

Compute derivative/integral of audio stream.

Applying both filters one after another produces original audio.