FFMPEG-PROTOCOLS(1)					   FFMPEG-PROTOCOLS(1)

NAME
       ffmpeg-protocols - FFmpeg protocols

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

PROTOCOL OPTIONS
       The libavformat library provides some generic global options, which can
       be set on all the protocols. In addition each protocol may support so-
       called private options, which are specific for that component.

       Options may be set by specifying -option value in the FFmpeg tools, or
       by setting the value explicitly in the "AVFormatContext" options or
       using the libavutil/opt.h API for programmatic use.

       The list of supported options follows:

       protocol_whitelist list (input)
	   Set a ","-separated list of allowed protocols. "ALL" matches all
	   protocols. Protocols prefixed by "-" are disabled.  All protocols
	   are allowed by default but protocols used by an another protocol
	   (nested protocols) are restricted to a per protocol subset.

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

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

       All protocols accept the following options:

       rw_timeout
	   Maximum time to wait for (network) read/write operations to
	   complete, in microseconds.

       A description of the currently available protocols follows.

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

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

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

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

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

       Muliple subscribers may stream from the broker using the command:

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

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

       The following options are supported:

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

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

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

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

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

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

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

   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.

       The accepted options are:

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

       URL Syntax is

	       cache:<URL>

   concat
       Physical concatenation protocol.

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

       A URL accepted by this protocol has the syntax:

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

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

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

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

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

   concatf
       Physical concatenation protocol using a line break delimited list of
       resources.

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

       A URL accepted by this protocol has the syntax:

	       concatf:<URL>

       where URL is the url containing a line break delimited list of
       resources to be concatenated, each one possibly specifying a distinct
       protocol. Special characters must be escaped with backslash or single
       quotes. See the "Quoting and escaping" section in the ffmpeg-utils(1)
       manual.

       For example to read a sequence of files split1.mpeg, split2.mpeg,
       split3.mpeg listed in separate lines within a file split.txt with
       ffplay use the command:

	       ffplay concatf:split.txt

       Where split.txt contains the lines:

	       split1.mpeg
	       split2.mpeg
	       split3.mpeg

   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:<URL>
	       crypto+<URL>

   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

   fd
       File descriptor access protocol.

       The accepted syntax is:

	       fd: -fd <file_descriptor>

       If fd is not specified, by default the stdout file descriptor will be
       used for writing, stdin for reading. Unlike the pipe protocol, fd
       protocol has seek support if it corresponding to a regular file. fd
       protocol doesn't support pass file descriptor via URL for security.

       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.

       fd  Set file descriptor.

   file
       File access protocol.

       Read from or write to a file.

       A file URL can have the form:

	       file:<filename>

       where filename is the path of the file to read.

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

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

	       ffmpeg -i 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.

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

       seekable
	   Controls if seekability is advertised on the file. 0 means non-
	   seekable, -1 means auto (seekable for normal files, non-seekable
	   for named pipes).

	   Many demuxers handle seekable and non-seekable resources
	   differently, overriding this might speed up opening certain files
	   at the cost of losing some features (e.g. accurate seeking).

   ftp
       FTP (File Transfer Protocol).

       Read from or write to remote resources using FTP protocol.

       Following syntax is required.

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

       This protocol accepts the following options.

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

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

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

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

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

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

   gopher
       Gopher protocol.

   gophers
       Gophers protocol.

       The Gopher protocol with TLS encapsulation.

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

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

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

   http
       HTTP (Hyper Text Transfer Protocol).

       This protocol accepts the following options:

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

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

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

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

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

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

       post_data
	   Set custom HTTP post data.

       referer
	   Set the Referer header. Include 'Referer: URL' header in HTTP
	   request.

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

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

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

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

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

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

       mime_type
	   Export the MIME type.

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

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

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

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

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

       offset
	   Set initial byte offset.

       end_offset
	   Try to limit the request to bytes preceding this offset.

       method
	   When used as a client option it sets the HTTP method for the
	   request.

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

       listen
	   If set to 1 enables experimental HTTP server. This can be used to
	   send data when used as an output option, or read data from a client
	   with HTTP POST when used as an input option.  If set to 2 enables
	   experimental multi-client HTTP server. This is not yet implemented
	   in ffmpeg.c and thus must not be used as a command line option.

		   # Server side (sending):
		   ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>

		   # Client side (receiving):
		   ffmpeg -i http://<server>:<port> -c copy somefile.ogg

		   # Client can also be done with wget:
		   wget http://<server>:<port> -O somefile.ogg

		   # Server side (receiving):
		   ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg

		   # Client side (sending):
		   ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>

		   # Client can also be done with wget:
		   wget --post-file=somefile.ogg http://<server>:<port>

       send_expect_100
	   Send an Expect: 100-continue header for POST. If set to 1 it will
	   send, if set to 0 it won't, if set to -1 it will try to send if it
	   is applicable. Default value is -1.

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

	   none
	       Choose the HTTP authentication type automatically. This is the
	       default.

	   basic
	       Choose the HTTP basic authentication.

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

       HTTP Cookies

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

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

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

   Icecast
       Icecast protocol (stream to Icecast servers)

       This protocol accepts the following options:

       ice_genre
	   Set the stream genre.

       ice_name
	   Set the stream name.

       ice_description
	   Set the stream description.

       ice_url
	   Set the stream website URL.

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

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

       password
	   Set the Icecast mountpoint password.

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

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

       tls Establish a TLS (HTTPS) connection to Icecast.

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

   ipfs
       InterPlanetary File System (IPFS) protocol support. One can access
       files stored on the IPFS network through so-called gateways. These are
       http(s) endpoints.  This protocol wraps the IPFS native protocols
       (ipfs:// and ipns://) to be sent to such a gateway. Users can (and
       should) host their own node which means this protocol will use one's
       local gateway to access files on the IPFS network.

       This protocol accepts the following options:

       gateway
	   Defines the gateway to use. When not set, the protocol will first
	   try locating the local gateway by looking at $IPFS_GATEWAY,
	   $IPFS_PATH and "$HOME/.ipfs/", in that order.

       One can use this protocol in 2 ways. Using IPFS:

	       ffplay ipfs://<hash>

       Or the IPNS protocol (IPNS is mutable IPFS):

	       ffplay ipns://<hash>

   mmst
       MMS (Microsoft Media Server) protocol over TCP.

   mmsh
       MMS (Microsoft Media Server) protocol over HTTP.

       The required syntax is:

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

   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.

	       # 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 to
       be seekable, so they will fail with the MD5 output protocol.

   pipe
       UNIX pipe access protocol.

       Read and write from UNIX pipes.

       The accepted syntax is:

	       pipe:[<number>]

       If fd isn't specified, 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.

       fd  Set file descriptor.

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

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

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

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

       The required syntax is:

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

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

       This protocol accepts the following options:

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

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

       Example usage:

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

   rist
       Reliable Internet Streaming Transport protocol

       The accepted options are:

       rist_profile
	   Supported values:

	   simple
	   main
	       This one is default.

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

       fifo_size
	   Size of the librist receiver output fifo in number of packets. This
	   must be a power of 2.  Defaults to 8192 (vs the librist default of
	   1024).

       overrun_nonfatal=1|0
	   Survive in case of librist fifo buffer overrun. Default value is 0.

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

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

       secret
	   Set override of encryption secret, by default is unset.

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

   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:

	       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_enhanced_codecs
	   Specify the list of codecs the client advertises to support in an
	   enhanced RTMP stream. This option should be set to a comma
	   separated list of fourcc values, like "hvc1,av01,vp09" for multiple
	   codecs or "hvc1" for only one codec. The specified list will be
	   presented in the "fourCcLive" property of the Connect Command
	   Message.

       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.

       tcp_nodelay=1|0
	   Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

	   Remark: Writing to the socket is currently not optimized to
	   minimize system calls and reduces the efficiency / effect of
	   TCP_NODELAY.

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

	       ffplay rtmp://myserver/vod/sample

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

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

   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.

   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.

   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.

   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.

   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.

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

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

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

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

   libssh
       Secure File Transfer Protocol via libssh

       Read from or write to remote resources using SFTP protocol.

       Following syntax is required.

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

       This protocol accepts the following options.

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

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

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

       Example: Play a file stored on remote server.

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

   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 RTMP
       protocol.

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

       The required syntax is:

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

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

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

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

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

       To play the same stream using ffplay:

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

   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.

       buffer_size=size
	   Set the maximum UDP socket buffer size in bytes.

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

       sources=ip[,ip]
	   List allowed source IP addresses.

       block=ip[,ip]
	   List disallowed (blocked) source IP addresses.

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

       localport=n
	   Set the local RTP port to n.

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

       timeout=n
	   Set timeout (in microseconds) of socket I/O operations to n.

	   This is a deprecated option. Instead, localrtpport should be used.

       Important notes:

       1.  If rtcpport is not set the RTCP port will be set to the RTP port
	   value plus 1.

       2.  If localrtpport (the local RTP port) is not set any available port
	   will be used for the local RTP and RTCP ports.

       3.  If localrtcpport (the local RTCP port) is not set it will be set to
	   the local RTP port value plus 1.

   rtsp
       Real-Time Streaming Protocol.

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

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

       The required syntax for a RTSP url is:

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

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

       Muxer

       The following options are supported.

       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.

	   Default value is 0.

       rtsp_flags
	   Set RTSP flags.

	   The following values are accepted:

	   latm
	       Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.

	   rfc2190
	       Use RFC 2190 packetization instead of RFC 4629 for H.263.

	   skip_rtcp
	       Don't send RTCP sender reports.

	   h264_mode0
	       Use mode 0 for H.264 in RTP.

	   send_bye
	       Send RTCP BYE packets when finishing.

	   Default value is 0.

       min_port
	   Set minimum local UDP port. Default value is 5000.

       max_port
	   Set maximum local UDP port. Default value is 65000.

       buffer_size
	   Set the maximum socket buffer size in bytes.

       pkt_size
	   Set max send packet size (in bytes). Default value is 1472.

       Demuxer

       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.

	   https
	       Use HTTPs tunneling as lower transport protocol, which is
	       useful for passing proxies and widely used for security
	       consideration.

	   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.

	   satip_raw
	       Export raw MPEG-TS stream instead of demuxing. The flag will
	       simply write out the raw stream, with the original PAT/PMT/PIDs
	       intact.

	   Default value is none.

       allowed_media_types
	   Set media types to accept from the server.

	   The following flags are accepted:

	   video
	   audio
	   data
	   subtitle

	   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.

       listen_timeout
	   Set maximum timeout (in seconds) to establish an initial
	   connection. Setting listen_timeout > 0 sets rtsp_flags to listen.
	   Default is -1 which means an infinite timeout when listen mode is
	   set.

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

       timeout
	   Set socket TCP I/O timeout in microseconds.

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

       buffer_size
	   Set the maximum socket buffer size in bytes.

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

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

       Examples

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

       o   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

       o   Watch a stream tunneled over HTTP:

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

       o   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

       o   Receive a stream in realtime:

		   ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

   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.

       Muxer

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

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

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

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

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

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

       same_port=0|1
	   If set to 1, send all RTP streams on the same port pair. If zero
	   (the default), all streams are sent on unique ports, with each
	   stream on a port 2 numbers higher than the previous.  VLC/Live555
	   requires this to be set to 1, to be able to receive the stream.
	   The RTP stack in libavformat for receiving requires all streams to
	   be sent on unique ports.

       Example command lines follow.

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

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

       Similarly, for watching in ffplay:

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

       And for watching in ffplay, over IPv6:

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

       Demuxer

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

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

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

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

       Example command lines follow.

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

	       ffplay sap://

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

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

   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.

   srt
       Haivision Secure Reliable Transport Protocol via libsrt.

       The supported syntax for a SRT URL is:

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

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

       or

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

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

       This protocol accepts the following options.

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

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

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

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

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

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

       listen_timeout=microseconds
	   Set socket listen timeout.

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

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

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

       nakreport=1|0
	   If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
	   periodically until a lost packet is retransmitted or intentionally
	   dropped. Default value is 1.

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

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

       enforced_encryption=1|0
	   If true, both connection parties must have the same password set
	   (including empty, that is, with no encryption). If the password
	   doesn't match or only one side is unencrypted, the connection is
	   rejected. Default is true.

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

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

       snddropdelay=microseconds
	   The sender's extra delay before dropping packets. This delay is
	   added to the default drop delay time interval value.

	   Special value -1: Do not drop packets on the sender at all.

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

       pkt_size=bytes
	   Alias for payload_size.

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

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

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

       recv_buffer_size=bytes
	   Set UDP receive buffer size, expressed in bytes.

       send_buffer_size=bytes
	   Set UDP send buffer size, expressed in bytes.

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

       tlpktdrop=1|0
	   Too-late Packet Drop. When enabled on receiver, it skips missing
	   packets that have not been delivered in time and delivers the
	   following packets to the application when their time-to-play has
	   come. It also sends a fake ACK to the sender. When enabled on
	   sender and enabled on the receiving peer, the sender drops the
	   older packets that have no chance of being delivered in time. It
	   was automatically enabled in the sender if the receiver supports
	   it.

       sndbuf=bytes
	   Set send buffer size, expressed in bytes.

       rcvbuf=bytes
	   Set receive buffer size, expressed in bytes.

	   Receive buffer must not be greater than ffs.

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

       minversion
	   The minimum SRT version that is required from the peer. A
	   connection to a peer that does not satisfy the minimum version
	   requirement will be rejected.

	   The version format in hex is 0xXXYYZZ for x.y.z in human readable
	   form.

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

       srt_streamid=string
	   Alias for streamid to avoid conflict with ffmpeg command line
	   option.

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

       messageapi=1|0
	   When set, this socket uses the Message API, otherwise it uses
	   Buffer API. Note that in live mode (see transtype) thereXs only
	   message API available. In File mode you can chose to use one of two
	   modes:

	   Stream API (default, when this option is false). In this mode you
	   may send as many data as you wish with one sending instruction, or
	   even use dedicated functions that read directly from a file. The
	   internal facility will take care of any speed and congestion
	   control. When receiving, you can also receive as many data as
	   desired, the data not extracted will be waiting for the next call.
	   There is no boundary between data portions in the Stream mode.

	   Message API. In this mode your single sending instruction passes
	   exactly one piece of data that has boundaries (a message). Contrary
	   to Live mode, this message may span across multiple UDP packets and
	   the only size limitation is that it shall fit as a whole in the
	   sending buffer. The receiver shall use as large buffer as necessary
	   to receive the message, otherwise the message will not be given up.
	   When the message is not complete (not all packets received or there
	   was a packet loss) it will not be given up.

       transtype=live|file
	   Sets the transmission type for the socket, in particular, setting
	   this option sets multiple other parameters to their default values
	   as required for a particular transmission type.

	   live: Set options as for live transmission. In this mode, you
	   should send by one sending instruction only so many data that fit
	   in one UDP packet, and limited to the value defined first in
	   payload_size (1316 is default in this mode). There is no speed
	   control in this mode, only the bandwidth control, if configured, in
	   order to not exceed the bandwidth with the overhead transmission
	   (retransmitted and control packets).

	   file: Set options as for non-live transmission. See messageapi for
	   further explanations

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

       tsbpd=1|0
	   When true, use Timestamp-based Packet Delivery mode. The default
	   behavior depends on the transmission type: enabled in live mode,
	   disabled in file mode.

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

   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.  If set to 0,
	   extract till end of file.

       Examples:

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

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

       Play an AVI file directly from a TAR archive:

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

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

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

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

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

   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 form key=val.

       The list of supported options follows.

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

       local_addr=addr
	   Local IP address of a network interface used for tcp socket
	   connect.

       local_port=port
	   Local port used for tcp socket connect.

       timeout=microseconds
	   Set raise error timeout, expressed in microseconds.

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

       listen_timeout=milliseconds
	   Set listen timeout, expressed in milliseconds.

       recv_buffer_size=bytes
	   Set receive buffer size, expressed bytes.

       send_buffer_size=bytes
	   Set send buffer size, expressed bytes.

       tcp_nodelay=1|0
	   Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

	   Remark: Writing to the socket is currently not optimized to
	   minimize system calls and reduces the efficiency / effect of
	   TCP_NODELAY.

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

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

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

   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 "AVOption"s):

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

       tls_verify=1|0
	   If enabled, try to verify the peer that we are communicating with.
	   Note, if using OpenSSL, this currently only makes sure that the
	   peer certificate is signed by one of the root certificates in the
	   CA database, but it does not validate that the certificate actually
	   matches the host name we are trying to connect to. (With other
	   backends, the host name is validated as well.)

	   This is disabled by default since it requires a CA database to be
	   provided by the caller in many cases.

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

       key_file, key=filename
	   A file containing the private key for the certificate.

       listen=1|0
	   If enabled, listen for connections on the provided port, and assume
	   the server role in the handshake instead of the client role.

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

       Example command lines:

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

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

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

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

   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 used
       to store the incoming data, which allows one to reduce loss of data due
       to UDP socket buffer overruns. The fifo_size and overrun_nonfatal
       options are related to this buffer.

       The list of supported options follows.

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

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

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

       localport=port
	   Override the local UDP port to bind with.

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

       pkt_size=size
	   Set the size in bytes of UDP packets.

       reuse=1|0
	   Explicitly allow or disallow reusing UDP sockets.

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

       connect=1|0
	   Initialize the UDP socket with "connect()". In this case, the
	   destination address can't be changed with ff_udp_set_remote_url
	   later.  If the destination address isn't known at the start, this
	   option can be specified in ff_udp_set_remote_url, too.  This allows
	   finding out the source address for the packets with getsockname,
	   and makes writes return with AVERROR(ECONNREFUSED) if "destination
	   unreachable" is received.  For receiving, this gives the benefit of
	   only receiving packets from the specified peer address/port.

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

       block=address[,address]
	   Ignore packets sent from the specified addresses. In case of
	   multicast, also exclude the source addresses in the multicast
	   subscription.

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

       overrun_nonfatal=1|0
	   Survive in case of UDP receiving circular buffer overrun. Default
	   value is 0.

       timeout=microseconds
	   Set raise error timeout, expressed in microseconds.

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

       broadcast=1|0
	   Explicitly allow or disallow UDP broadcasting.

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

       Examples

       o   Use ffmpeg to stream over UDP to a remote endpoint:

		   ffmpeg -i <input> -f <format> udp://<hostname>:<port>

       o   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

       o   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 URL is:

	       unix://<filepath>

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

       timeout
	   Timeout in ms.

       listen
	   Create the Unix socket in listening mode.

   zmq
       ZeroMQ asynchronous messaging using the libzmq library.

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

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

	       zmq:tcp://ip-address:port

       Example: Create a localhost stream on port 5555:

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

       Multiple clients may connect to the stream using:

	       ffplay zmq:tcp://127.0.0.1:5555

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

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

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

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

SEE ALSO
       ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)

AUTHORS
       The FFmpeg developers.

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

       Maintainers for the specific components are listed in the file
       MAINTAINERS in the source code tree.

							   FFMPEG-PROTOCOLS(1)
