FFmpeg Protocols Documentation

Table of Contents

• 1 Description
• 2 Protocols
  • 2.1 bluray
  • 2.2 cache
  • 2.3 concat
  • 2.4 crypto
  • 2.5 data
  • 2.6 file
  • 2.7 ftp
  • 2.8 gopher
  • 2.9 hls
  • 2.10 http
    • 2.10.1 HTTP Cookies
  • 2.11 Icecast
  • 2.12 mmst
  • 2.13 mmsh
  • 2.14 md5
  • 2.15 pipe
  • 2.16 rtmp
  • 2.17 rtmpe
  • 2.18 rtmps
  • 2.19 rtmpt
  • 2.20 rtmpte
  • 2.21 rtmpts
  • 2.22 libsmbclient
  • 2.23 libssh
  • 2.24 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
  • 2.25 rtp
  • 2.26 rtsp
    • 2.26.1 Examples
  • 2.27 sap
    • 2.27.1 Muxer
    • 2.27.2 Demuxer
  • 2.28 sctp
  • 2.29 srtp
  • 2.30 subfile
  • 2.31 tcp
  • 2.32 tls
  • 2.33 udp
    • 2.33.1 Examples
  • 2.34 unix
• 3 See Also
• 4 Authors

1 Description

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

2 Protocols

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

When you configure your FFmpeg build, all the supported protocols areenabled by default. You can list all available ones using theconfigure option "–list-protocols".

You can disable all the protocols using the configure option"–disable-protocols", and selectively enable a protocol using theoption "–enable-protocol=PROTOCOL", or you can disable aparticular protocol using the option"–disable-protocol=PROTOCOL".

The option "-protocols" of the ff* tools will display the list ofsupported protocols.

A description of the currently available protocols follows.

2.1 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

2.2 cache

Caching wrapper for input stream.

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

cache:URL

2.3 concat

Physical concatenation protocol.

Allow to read and seek from many resource in sequence as if they werea unique resource.

A URL accepted by this protocol has the syntax:

concat:URL1|URL2|...|URLN

where URL1, URL2, ..., URLN are the urls of theresource to be concatenated, each one possibly specifying a distinctprotocol.

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

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

Note that you may need to escape the character "|" which is special formany shells.

2.4 crypto

AES-encrypted stream reading protocol.

The accepted options are:

key

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

iv

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

Accepted URL formats:

crypto:URLcrypto+URL

2.5 data

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

2.6 file

File access protocol.

Allow to 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 afile URL. Depending on the build, an URL that looks like a Windowspath with the drive letter at the beginning will also be assumed to bea file URL (usually not the case in builds for unix-like systems).

For example to read from a file input.mpeg with ffmpeguse 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 preventstruncating. Default value is 1.

blocksize

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

2.7 ftp

FTP (File Transfer Protocol).

Allow to 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 leveloperation. By default it is set to -1, which means that the timeout isnot specified.

ftp-anonymous-password

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

ftp-write-seekable

Control seekability of connection during encoding. If set to 1 theresource is supposed to be seekable, if set to 0 it is assumed notto be seekable. Default value is 0.

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

2.8 gopher

Gopher protocol.

2.9 hls

Read Apple HTTP Live Streaming compliant segmented stream asa uniform one. The M3U8 playlists describing the segments can beremote HTTP resources or local files, accessed using the standardfile protocol.The nested protocol is declared by specifying"+proto" after the hls URI scheme name, where protois either "file" or "http".

hls+http://host/path/to/remote/resource.m3u8hls+file://path/to/local/resource.m3u8

Using this protocol is discouraged - the hls demuxer should workjust as well (if not, please report the issues) and is more complete.To use the hls demuxer instead, simply use the direct URLs to them3u8 files.

2.10 http

HTTP (Hyper Text Transfer Protocol).

This protocol accepts the following options:

seekable

Control seekability of connection. If set to 1 the resource issupposed 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. Defaultvalue 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. Thevalue 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 astring describing the libavformat build. ("Lavf/<version>")

timeout

Set timeout in microseconds of socket I/O operations used by the underlying low leveloperation. By default it is set to -1, which means that the timeout isnot specified.

mime_type

Export the MIME type.

icy

If set to 1 request ICY (SHOUTcast) metadata from the server. If the serversupports this, the metadata has to be retrieved by the application by readingtheicy_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 replyheaders, separated by newline characters.

icy_metadata_packet

If the server supports ICY metadata, and icy was set to 1, thiscontains the last non-empty metadata packet sent by the server. It should bepolled in regular intervals by applications interested in mid-stream metadataupdates.

cookies

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

offset

Set initial byte offset.

end_offset

Try to limit the request to bytes preceding this offset.

2.10.1 HTTP Cookies

Some HTTP requests will be denied unless cookie values are passed in with therequest. Thecookies option allows these cookies to be specified. Atthe 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 thecookie value in the HTTP Cookie header field. Multiple cookies can be delimitedby 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

2.11 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 fromaudio/mpeg.

legacy_icecast

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

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

2.12 mmst

MMS (Microsoft Media Server) protocol over TCP.

2.13 mmsh

MMS (Microsoft Media Server) protocol over HTTP.

The required syntax is:

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

2.14 md5

MD5 output protocol.

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

Some examples follow.

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

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

2.15 pipe

UNIX pipe access protocol.

Allow to read and write from UNIX pipes.

The accepted syntax is:

pipe:[number]

number is the number corresponding to the file descriptor of thepipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). Ifnumberis not specified, by default the stdout file descriptor will be usedfor 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 isINT_MAX, which results in not limiting the requested block size.Setting this value reasonably low improves user termination request reactiontime, which is valuable if data transmission is slow.

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

2.16 rtmp

Real-Time Messaging Protocol.

The Real-Time Messaging Protocol (RTMP) is used for streaming multimediacontent 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 tothe path where the application is installed on the RTMP server(e.g./ondemand/, /flash/live/, etc.). You can overridethe value parsed from the URI through thertmp_app option, too.

playpath

It is the path or name of the resource to play with reference to theapplication specified inapp, may be prefixed by "mp4:". Youcan override the value parsed from the URI through thertmp_playpathoption, 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 viaAVOptions):

rtmp_app

Name of application to connect on the RTMP server. This optionoverrides 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 forFALSE or TRUE, respectively. Likewise for Objects the data must be 0 or1 to end or begin an object, respectively. Data items in subobjects maybe named, by prefixing the type with ’N’ and specifying the name beforethe value (i.e.NB:myFlag:1). This option may be used multipletimes to construct arbitrary AMF sequences.

rtmp_flashver

Version of the Flash plugin used to run the SWF player. The defaultis 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 defaultis 10.

rtmp_live

Specify that the media is a live stream. No resuming or seeking inlive streams is possible. The default value isany, which means thesubscriber first tries to play the live stream specified in theplaypath. If a live stream of that name is not found, it plays therecorded stream. The other possible values arelive andrecorded.

rtmp_pageurl

URL of the web page in which the media was embedded. By default novalue will be sent.

rtmp_playpath

Stream identifier to play or to publish. This option overrides theparameter 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_liveis 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 andapp names separately:

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

2.17 rtmpe

Encrypted Real-Time Messaging Protocol.

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

2.18 rtmps

Real-Time Messaging Protocol over a secure SSL connection.

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

2.19 rtmpt

Real-Time Messaging Protocol tunneled through HTTP.

The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is usedfor streaming multimedia content within HTTP requests to traversefirewalls.

2.20 rtmpte

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 traversefirewalls.

2.21 rtmpts

Real-Time Messaging Protocol tunneled through HTTPS.

The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is usedfor streaming multimedia content within HTTPS requests to traversefirewalls.

2.22 libsmbclient

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 miliseconds of socket I/O operations used by the underlyinglow level operation. By default it is set to -1, which means that the timeoutis not specified.

truncate

Truncate existing files on write, if set to 1. A value of 0 preventstruncating. 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/.

2.23 libssh

Secure File Transfer Protocol via libssh

Allow to 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 leveloperation. By default it is set to -1, which means that the timeoutis not specified.

truncate

Truncate existing files on write, if set to 1. A value of 0 preventstruncating. 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

2.24 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte

Real-Time Messaging Protocol and its variants supported throughlibrtmp.

Requires the presence of the librtmp headers and library duringconfiguration. You need to explicitly configure the build with"–enable-librtmp". If enabled this will replace the native RTMPprotocol.

This protocol provides most client functions and a few serverfunctions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneledvariants 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, andserver,port, app and playpath have the samemeaning as specified for the RTMP native protocol.options contains a list of space-separated options of the formkey=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 usingffmpeg:

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

To play the same stream using ffplay:

ffplay "rtmp://myserver/live/mystream live=1"

2.25 rtp

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.

connect=0|1

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

Important notes:

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

2.26 rtsp

Real-Time Streaming Protocol.

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

The muxer can be used to send a stream using RTSP ANNOUNCE to a serversupporting it (currently Darwin Streaming Server and Mischa Spiegelmock’sRTSP server).

The required syntax for a RTSP url is:

rtsp://hostname[:port]/path

Options can be set on the ffmpeg/ffplay commandline, or set in code viaAVOptions or inavformat_open_input.

The following options are supported.

initial_pause

Do not start playing the stream immediately if set to 1. Default valueis 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 lowertransport protocol.

‘udp_multicast’

Use UDP multicast as lower transport protocol.

‘http’

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

Multiple lower transport protocols may be specified, in that case they aretried 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 thertsp_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 thelibavformat 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). Thiscan be disabled by setting the maximum demuxing delay to zero (viathemax_delay field of AVFormatContext).

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

2.26.1 Examples

The following examples all make use of the ffplay andffmpeg 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

2.27 sap

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

2.27.1 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 optionsare supported:

announce_addr=address

Specify the destination IP address for sending the announcements to.If omitted, the announcements are sent to the commonly used SAPannouncement multicast address 224.2.127.254 (sap.mcast.net), orff0e::2:7ffe ifdestination is an IPv6 address.

announce_port=port

Specify the port to send the announcements on, defaults to9875 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 (thedefault), all streams are sent on unique ports, with each stream on aport 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 senton 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]

2.27.2 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.portis 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]

2.28 sctp

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.

2.29 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 abase64-encoded representation of a binary block. The first 16 bytes ofthis binary block are used as master key, the following 14 bytes areused as master salt.

2.30 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 DVD VOB file (start and end sectors obtainedexternally 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

2.31 tcp

Transmission Control Protocol.

The required syntax for a TCP url is:

tcp://hostname:port[?options]

options contains a list of &-separated options of the formkey=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 morethan this time interval, raise error.

listen_timeout=microseconds

Set listen timeout, expressed in microseconds.

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

ffmpeg -i input -f format tcp://hostname:port?listenffplay tcp://hostname:port

2.32 tls

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 AVOptions):

ca_file, cafile=filename

A file containing certificate authority (CA) root certificates to treatas trusted. If the linked TLS library contains a default this might notneed to be specified for verification to work, but not all libraries andsetups 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 thepeer certificate is signed by one of the root certificates in the CAdatabase, but it does not validate that the certificate actuallymatches 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 beprovided 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 requiredby the peer, while client certificates only are mandated in certainsetups.)

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 assumethe server role in the handshake instead of the client role.

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

2.33 udp

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 usedto store the incoming data, which allows one to reduce loss of data due toUDP socket buffer overruns. Thefifo_size andoverrun_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 eitherthe receive or send buffer size, depending on what the socket is used for.Default is 64KB. See alsofifo_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 multicastand the host has multiple interfaces, where the user can choosewhich 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, thedestination address can’t be changed with ff_udp_set_remote_url later.If the destination address isn’t known at the start, this option canbe 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 "destinationunreachable" is received.For receiving, this gives the benefit of only receiving packets fromthe specified peer address/port.

sources=address[,address]

Only receive packets sent to the multicast group from one of thespecified sender IP addresses.

block=address[,address]

Ignore packets sent to the multicast group from the specifiedsender IP addresses.

fifo_size=units

Set the UDP receiving circular buffer size, expressed as a number ofpackets 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. Defaultvalue is 0.

timeout=microseconds

Set raise error timeout, expressed in microseconds.

This option is only relevant in read mode: if no data arrived in morethan this time interval, raise error.

broadcast=1|0

Explicitly allow or disallow UDP broadcasting.

Note that broadcasting may not work properly on networks havinga broadcast storm protection.

2.33.1 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 188sized 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 ...

2.34 unix

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 AVOptions):

timeout

Timeout in ms.

listen

Create the Unix socket in listening mode.

3 See Also

ffmpeg, ffplay, ffprobe,ffserver,libavformat