~siretart/libav/trusty

@chapter Protocols

@c man begin PROTOCOLS

Protocols are configured elements in Libav which allow to access

resources which require the use of a particular protocol.

When you configure your Libav 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 av* tools will display the list of

supported protocols.

A description of the currently available protocols follows.

@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{avplay} use the

command:

@example

avplay 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 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{avconv}

use the command:

@example

avconv -i file:input.mpeg output.mpeg

@end example

The av* 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).

@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.

avconv -i input.flv -f avi -y md5:output.avi.md5

# Write the MD5 hash of the encoded AVI file to stdout.

avconv -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{avconv}:

@example

cat test.wav | avconv -i pipe:0

# ...this is the same as...

cat test.wav | avconv -i pipe:

@end example

For writing to stdout with @command{avconv}:

@example

avconv -i test.wav -f avi pipe:1 | cat > test.avi

# ...this is the same as...

avconv -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{avplay} a multimedia resource named

"sample" from the application "vod" from an RTMP server "myserver":

@example

avplay 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{avconv}:

@example

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

@end example

To play the same stream using @command{avplay}:

@example

avplay "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{avconv}/@command{avplay} 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{avplay}, 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

avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

@end example

To watch a stream tunneled over HTTP:

@example

avplay -rtsp_transport http rtsp://server/video.mp4

@end example

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

@example

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

@end example

To receive a stream in realtime:

@example

avconv -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

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

@end example

Similarly, for watching in avplay:

@example

avconv -re -i @var{input} -f sap sap://224.0.0.255

@end example

And for watching in avplay, over IPv6:

@example

avconv -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

avplay sap://

@end example

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

@example

avplay 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

@example

avconv -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen

avplay tcp://@var{hostname}:@var{port}

@end example

@end table

@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 &-seperated options of the form @var{key}=@var{val}.

Follow the list of supported options.

@table @option

@item buffer_size=@var{size}

set the UDP buffer size in bytes

@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.

@end table

Some usage examples of the udp protocol with @command{avconv} follow.

To stream over UDP to a remote endpoint:

@example

avconv -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

avconv -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535