ffmpeg-protocols (1)

Leading comments

Automatically generated by Pod::Man 4.09 (Pod::Simple 3.35)

Standard preamble:
========================================================================

(The comments found at the beginning of the groff file "man1/ffmpeg-protocols.1".)

NAME

ffmpeg-protocols - FFmpeg protocols

DESCRIPTION

This document describes the input and output protocols provided by the libavformat library.

PROTOCOLS

Protocols are configured elements in FFmpeg that enable access to resources that require specific protocols.

When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list all available ones using the configure option ``--list-protocols''.

You can disable all the protocols using the configure option ``--disable-protocols'', and selectively enable a protocol using the option "--enable-protocol=

``, or you can disable a particular protocol using the option ''--disable-protocol=".

The option ``-protocols'' of the ff* tools will display the list of supported protocols.

A description of the currently available protocols follows.

async

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

bluray

Read BluRay playlist.

The accepted options are:

angle

BluRay angle

chapter

Start chapter (1...N)

playlist

Playlist to read (

BDMV/PLAYLIST/

?????.mpls)

Examples:

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

        bluray:/mnt/bluray

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

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

cache

Caching wrapper for input stream.

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

        cache:<URL>

concat

Physical concatenation protocol.

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

A

accepted by this protocol has the syntax:

        concat:<URL1>|<URL2>|...|<URLN>

where

, , ..., 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.

crypto

AES-encrypted stream reading protocol.

The accepted options are:

key

Set the

AES

decryption key binary block from given hexadecimal representation.

iv

Set the

AES

decryption initialization vector binary block from given hexadecimal representation.

Accepted

formats:

        crypto:<URL>
        crypto+<URL>

data

Data in-line in the See <en.wikipedia.org/wiki/Data_URI_scheme>.

For example, to convert a

file given inline with ffmpeg:

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

file

File access protocol.

Read from or write to a file.

A file

can have the form:

        file:<filename>

where filename is the path of the file to read.

An

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

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

        ffmpeg -i file:input.mpeg output.mpeg

This protocol accepts the following options:

truncate

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

blocksize

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

ftp

(File Transfer Protocol).

Read from or write to remote resources using

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-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.

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 servers behave in different way during seek operation. ff* tools may produce incomplete content due to server limitations.

gopher

Gopher protocol.

hls

Read Apple Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying "+proto" after the hls 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.

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.

user-agent

user_agent

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

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.

mime_type

Export the

MIME

type.

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 mutli-client

HTTP

server. This is not yet implemented in ffmpeg.c or ffserver.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>

Cookies

Some

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. requests that match both the domain and path will automatically include the cookie value in the 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;" somedomain.com/somestream.m3u8

Icecast

Icecast protocol (stream to Icecast servers)

This protocol accepts the following options:

ice_genre

Set the stream genre.

ice_name

Set the stream name.

ice_description

Set the stream description.

ice_url

Set the stream website

URL.

ice_public

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

user_agent

Override the User-Agent header. If not specified a string of the form ``Lavf/<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.

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

mmst

(Microsoft Media Server) protocol over

mmsh

(Microsoft Media Server) protocol over

The required syntax is:

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

md5

output protocol.

Computes the

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

Some examples follow.

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

Note that some formats (typically

) require the output protocol to be seekable, so they will fail with the output protocol.

pipe

pipe access protocol.

Read and write from

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 | ffmpeg -i pipe:

For writing to stdout with ffmpeg:

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

This protocol accepts the following options:

blocksize

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

Note that some formats (typically

), require the output protocol to be seekable, so they will fail with the pipe output protocol.

rtmp

Real-Time Messaging Protocol.

The Real-Time Messaging Protocol (

) is used for streaming multimedia content across a 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

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/

rtmpe

Encrypted Real-Time Messaging Protocol.

The Encrypted Real-Time Messaging Protocol (

) is used for streaming multimedia content within standard cryptographic primitives, consisting of Diffie-Hellman key exchange and generating a pair of keys.

rtmps

Real-Time Messaging Protocol over a secure connection.

The Real-Time Messaging Protocol (

) is used for streaming multimedia content across an encrypted connection.

rtmpt

Real-Time Messaging Protocol tunneled through

The Real-Time Messaging Protocol tunneled through

() is used for streaming multimedia content within requests to traverse firewalls.

rtmpte

Encrypted Real-Time Messaging Protocol tunneled through

The Encrypted Real-Time Messaging Protocol tunneled through

() is used for streaming multimedia content within requests to traverse firewalls.

rtmpts

Real-Time Messaging Protocol tunneled through

The Real-Time Messaging Protocol tunneled through

() is used for streaming multimedia content within requests to traverse firewalls.

libsmbclient

libsmbclient permits one to manipulate network resources.

Following syntax is required.

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

This protocol accepts the following options.

timeout

Set timeout in miliseconds 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: <www.samba.org>.

libssh

Secure File Transfer Protocol via libssh

Read from or write to remote resources using

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

librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte

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

protocol.

This protocol provides most client functions and a few server functions needed to support

tunneled in (), encrypted (), over () and tunneled variants of these encrypted types ().

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

variant, and server, port, app and playpath have the same meaning as specified for the 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

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"

rtp

Real-time Transport Protocol.

The required syntax for an

is: rtp://hostname[:port][?option=val...]

port specifies the

port to use.

The following

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.

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.

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.

rtsp

Real-Time Streaming Protocol.

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

The muxer can be used to send a stream using

to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's <github.com/revmischa/rtsp-server>).

The required syntax for a

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

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>

sap

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

Muxer

The syntax for a

url given to the muxer is:

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

The

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

        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

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

multicast address:

        ffplay sap://

To play back the first stream announced on one the default IPv6

multicast address:

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

sctp

Stream Control Transmission Protocol.

The accepted

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.

srtp

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

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.

subfile

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.

Examples:

Extract a chapter from a

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

file directly from a archive:

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

tcp

Transmission Control Protocol.

The required syntax for a

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=1|0

Listen for an incoming connection. 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.

The following example shows how to setup a listening

connection with ffmpeg, which is then accessed with ffplay:

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

tls

Transport Layer Security () / Secure Sockets Layer ()

The required syntax for a

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 GnuTLS, 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.

Example command lines:

To create a

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

server using ffplay:

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

udp

User Datagram Protocol.

The required syntax for an

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

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 64KB. See also fifo_size.

localport=port

Override the local

UDP

port to bind with.

localaddr=addr

Choose the local

IP

address. This is useful e.g. if sending multicast and the host has multiple interfaces, where the user can choose which interface to send on by specifying the

IP

address of that interface.

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 to the multicast group from one of the specified sender

IP

addresses.

block=address[,address]

Ignore packets sent to the multicast group from the specified sender

IP

addresses.

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

Unix local socket

The required syntax for a Unix socket

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.

SEE ALSO

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

AUTHORS

The FFmpeg developers.

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

Maintainers for the specific components are listed in the file

in the source code tree.