FFmpeg/doc/protocols.texi

@chapter Protocols
@c man begin PROTOCOLS

Protocols are configured elements in FFmpeg which allow to access
resources which require the use of a particular protocol.

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=@var{PROTOCOL}", or you can disable a
particular protocol using the option
"--disable-protocol=@var{PROTOCOL}".

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

A description of the currently available protocols follows.

@section bluray

Read BluRay playlist.

The accepted options are:
@table @option

@item angle
BluRay angle

@item chapter
Start chapter (1...N)

@item playlist
Playlist to read (BDMV/PLAYLIST/?????.mpls)

@end table

Examples:

Read longest playlist from BluRay mounted to /mnt/bluray:
@example
bluray:/mnt/bluray
@end example

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

@section concat

Physical concatenation protocol.

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

A URL accepted by this protocol has the syntax:
@example
concat:@var{URL1}|@var{URL2}|...|@var{URLN}
@end example

where @var{URL1}, @var{URL2}, ..., @var{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 @file{split1.mpeg},
@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
command:
@example
ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
@end example

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

@section data

Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.

For example, to convert a GIF file given inline with @command{ffmpeg}:
@example
ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
@end example

@section file

File access protocol.

Allow to read from or read to a file.

For example to read from a file @file{input.mpeg} with @command{ffmpeg}
use the command:
@example
ffmpeg -i file:input.mpeg output.mpeg
@end example

The ff* tools default to the file protocol, that is a resource
specified with the name "FILE.mpeg" is interpreted as the URL
"file:FILE.mpeg".

@section gopher

Gopher protocol.

@section hls

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
"+@var{proto}" after the hls URI scheme name, where @var{proto}
is either "file" or "http".

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

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.

@section http

HTTP (Hyper Text Transfer Protocol).

This protocol accepts the following options.

@table @option
@item 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.

@item chunked_post
If set to 1 use chunked transfer-encoding for posts, default is 1.

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

@item content_type
Force a content type.

@item user-agent
Override User-Agent header. If not specified the protocol will use a
string describing the libavformat build.

@item multiple_requests
Use persistent connections if set to 1. By default it is 0.

@item post_data
Set custom HTTP post data.

@item 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.

@item mime_type
Set MIME type.
@end table

@section mmst

MMS (Microsoft Media Server) protocol over TCP.

@section mmsh

MMS (Microsoft Media Server) protocol over HTTP.

The required syntax is:
@example
mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
@end example

@section md5

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.
@example
# 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:
@end example

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

@section pipe

UNIX pipe access protocol.

Allow to read and write from UNIX pipes.

The accepted syntax is:
@example
pipe:[@var{number}]
@end example

@var{number} is the number corresponding to the file descriptor of the
pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).  If @var{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 @command{ffmpeg}:
@example
cat test.wav | ffmpeg -i pipe:0
# ...this is the same as...
cat test.wav | ffmpeg -i pipe:
@end example

For writing to stdout with @command{ffmpeg}:
@example
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
@end example

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

@section rtmp

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:
@example
rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
@end example

The accepted parameters are:
@table @option

@item server
The address of the RTMP server.

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

@item 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. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
the value parsed from the URI through the @code{rtmp_app} option, too.

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

@item listen
Act as a server, listening for an incoming connection.

@item timeout
Maximum time to wait for the incoming connection. Implies listen.
@end table

Additionally, the following parameters can be set via command line options
(or in code via @code{AVOption}s):
@table @option

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

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

@item rtmp_conn
Extra arbitrary AMF connection parameters, parsed from a string,
e.g. like @code{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. @code{NB:myFlag:1}). This option may be used multiple
times to construct arbitrary AMF sequences.

@item rtmp_flashver
Version of the Flash plugin used to run the SWF player. The default
is LNX 9,0,124,2.

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

@item rtmp_live
Specify that the media is a live stream. No resuming or seeking in
live streams is possible. The default value is @code{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 @code{live} and
@code{recorded}.

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

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

@item 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.

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

@item rtmp_swfsize
Size of the decompressed SWF file, required for SWFVerification.

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

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

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

@end table

For example to read with @command{ffplay} a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
@example
ffplay rtmp://myserver/vod/sample
@end example

@section rtmpe

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.

@section rtmps

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.

@section rtmpt

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.

@section 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 traverse
firewalls.

@section rtmpts

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.

@section 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 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:
@example
@var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
@end example

where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
@var{server}, @var{port}, @var{app} and @var{playpath} have the same
meaning as specified for the RTMP native protocol.
@var{options} contains a list of space-separated options of the form
@var{key}=@var{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
@command{ffmpeg}:
@example
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
@end example

To play the same stream using @command{ffplay}:
@example
ffplay "rtmp://myserver/live/mystream live=1"
@end example

@section rtp

Real-Time Protocol.

@section rtsp

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
@uref{http://github.com/revmischa/rtsp-server, RTSP server}).

The required syntax for a RTSP url is:
@example
rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example

The following options (set on the @command{ffmpeg}/@command{ffplay} command
line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
are supported:

Flags for @code{rtsp_transport}:

@table @option

@item udp
Use UDP as lower transport protocol.

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

@item udp_multicast
Use UDP multicast as lower transport protocol.

@item http
Use HTTP tunneling as lower transport protocol, which is useful for
passing proxies.
@end table

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 @code{tcp} and @code{udp} options are supported.

Flags for @code{rtsp_flags}:

@table @option
@item filter_src
Accept packets only from negotiated peer address and port.
@item listen
Act as a server, listening for an incoming connection.
@end table

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 @code{max_delay} field of AVFormatContext).

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

Example command lines:

To watch a stream over UDP, with a max reordering delay of 0.5 seconds:

@example
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
@end example

To watch a stream tunneled over HTTP:

@example
ffplay -rtsp_transport http rtsp://server/video.mp4
@end example

To send a stream in realtime to a RTSP server, for others to watch:

@example
ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example

To receive a stream in realtime:

@example
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
@end example

@section sap

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.

@subsection Muxer

The syntax for a SAP url given to the muxer is:
@example
sap://@var{destination}[:@var{port}][?@var{options}]
@end example

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

@table @option

@item announce_addr=@var{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 @var{destination} is an IPv6 address.

@item announce_port=@var{port}
Specify the port to send the announcements on, defaults to
9875 if not specified.

@item ttl=@var{ttl}
Specify the time to live value for the announcements and RTP packets,
defaults to 255.

@item same_port=@var{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.
@end table

Example command lines follow.

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

@example
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
@end example

Similarly, for watching in @command{ffplay}:

@example
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
@end example

And for watching in @command{ffplay}, over IPv6:

@example
ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
@end example

@subsection Demuxer

The syntax for a SAP url given to the demuxer is:
@example
sap://[@var{address}][:@var{port}]
@end example

@var{address} is the multicast address to listen for announcements on,
if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{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:

@example
ffplay sap://
@end example

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

@example
ffplay sap://[ff0e::2:7ffe]
@end example

@section tcp

Trasmission Control Protocol.

The required syntax for a TCP url is:
@example
tcp://@var{hostname}:@var{port}[?@var{options}]
@end example

@table @option

@item listen
Listen for an incoming connection

@item timeout=@var{microseconds}
In read mode: if no data arrived in more than this time interval, raise error.
In write mode: if socket cannot be written in more than this time interval, raise error.
This also sets timeout on TCP connection establishing.

@example
ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
ffplay tcp://@var{hostname}:@var{port}
@end example

@end table

@section tls

Transport Layer Security/Secure Sockets Layer

The required syntax for a TLS/SSL url is:
@example
tls://@var{hostname}:@var{port}[?@var{options}]
@end example

@table @option

@item listen
Act as a server, listening for an incoming connection.

@item cafile=@var{filename}
Certificate authority file. The file must be in OpenSSL PEM format.

@item cert=@var{filename}
Certificate file. The file must be in OpenSSL PEM format.

@item key=@var{filename}
Private key file.

@item verify=@var{0|1}
Verify the peer's certificate.

@end table

Example command lines:

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

@example
ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
@end example

To play back a stream from the TLS/SSL server using @command{ffplay}:

@example
ffplay tls://@var{hostname}:@var{port}
@end example

@section udp

User Datagram Protocol.

The required syntax for a UDP url is:
@example
udp://@var{hostname}:@var{port}[?@var{options}]
@end example

@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.

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

The list of supported options follows.

@table @option

@item buffer_size=@var{size}
Set the UDP socket buffer size in bytes. This is used both for the
receiving and the sending buffer size.

@item localport=@var{port}
Override the local UDP port to bind with.

@item localaddr=@var{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.

@item pkt_size=@var{size}
Set the size in bytes of UDP packets.

@item reuse=@var{1|0}
Explicitly allow or disallow reusing UDP sockets.

@item ttl=@var{ttl}
Set the time to live value (for multicast only).

@item connect=@var{1|0}
Initialize the UDP socket with @code{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.

@item sources=@var{address}[,@var{address}]
Only receive packets sent to the multicast group from one of the
specified sender IP addresses.

@item block=@var{address}[,@var{address}]
Ignore packets sent to the multicast group from the specified
sender IP addresses.

@item fifo_size=@var{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.

@item overrun_nonfatal=@var{1|0}
Survive in case of UDP receiving circular buffer overrun. Default
value is 0.

@item timeout=@var{microseconds}
In read mode: if no data arrived in more than this time interval, raise error.
@end table

Some usage examples of the UDP protocol with @command{ffmpeg} follow.

To stream over UDP to a remote endpoint:
@example
ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
@end example

To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
@example
ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
@end example

To receive over UDP from a remote endpoint:
@example
ffmpeg -i udp://[@var{multicast-address}]:@var{port}
@end example

@c man end PROTOCOLS