From 4a6d9312cc1c9d62d66c4def71246d9faae29edb Mon Sep 17 00:00:00 2001
From: zhangmeng <775834166@qq.com>
Date: 星期三, 03 三月 2021 14:26:16 +0800
Subject: [PATCH] environment settings
---
module/VideoPlayer/lib/linux/ffmpeg/share/man/man1/ffmpeg-protocols.1 | 1826 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
1 files changed, 1,826 insertions(+), 0 deletions(-)
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@@ -0,0 +1,1826 @@
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+.\" ========================================================================
+.\"
+.IX Title "FFMPEG-PROTOCOLS 1"
+.TH FFMPEG-PROTOCOLS 1 " " " " " "
+.\" For nroff, turn off justification. Always turn off hyphenation; it makes
+.\" way too many mistakes in technical documents.
+.if n .ad l
+.nh
+.SH "NAME"
+ffmpeg\-protocols \- FFmpeg protocols
+.SH "DESCRIPTION"
+.IX Header "DESCRIPTION"
+This document describes the input and output protocols provided by the
+libavformat library.
+.SH "PROTOCOL OPTIONS"
+.IX Header "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.
+.PP
+Options may be set by specifying \-\fIoption\fR \fIvalue\fR in the
+FFmpeg tools, or by setting the value explicitly in the
+\&\f(CW\*(C`AVFormatContext\*(C'\fR options or using the \fIlibavutil/opt.h\fR \s-1API\s0
+for programmatic use.
+.PP
+The list of supported options follows:
+.IP "\fBprotocol_whitelist\fR \fIlist\fR \fB(\fR\fIinput\fR\fB)\fR" 4
+.IX Item "protocol_whitelist list (input)"
+Set a \*(L",\*(R"\-separated list of allowed protocols. \*(L"\s-1ALL\s0\*(R" matches all protocols. Protocols
+prefixed by \*(L"\-\*(R" are disabled.
+All protocols are allowed by default but protocols used by an another
+protocol (nested protocols) are restricted to a per protocol subset.
+.SH "PROTOCOLS"
+.IX Header "PROTOCOLS"
+Protocols are configured elements in FFmpeg that enable access to
+resources that require specific protocols.
+.PP
+When you configure your FFmpeg build, all the supported protocols are
+enabled by default. You can list all available ones using the
+configure option \*(L"\-\-list\-protocols\*(R".
+.PP
+You can disable all the protocols using the configure option
+\&\*(L"\-\-disable\-protocols\*(R", and selectively enable a protocol using the
+option "\-\-enable\-protocol=\fI\s-1PROTOCOL\s0\fR\*(L", or you can disable a
+particular protocol using the option
+\&\*(R"\-\-disable\-protocol=\fI\s-1PROTOCOL\s0\fR".
+.PP
+The option \*(L"\-protocols\*(R" of the ff* tools will display the list of
+supported protocols.
+.PP
+All protocols accept the following options:
+.IP "\fBrw_timeout\fR" 4
+.IX Item "rw_timeout"
+Maximum time to wait for (network) read/write operations to complete,
+in microseconds.
+.PP
+A description of the currently available protocols follows.
+.SS "async"
+.IX Subsection "async"
+Asynchronous data filling wrapper for input stream.
+.PP
+Fill data in a background thread, to decouple I/O operation from demux thread.
+.PP
+.Vb 3
+\& async:<URL>
+\& async:http://host/resource
+\& async:cache:http://host/resource
+.Ve
+.SS "bluray"
+.IX Subsection "bluray"
+Read BluRay playlist.
+.PP
+The accepted options are:
+.IP "\fBangle\fR" 4
+.IX Item "angle"
+BluRay angle
+.IP "\fBchapter\fR" 4
+.IX Item "chapter"
+Start chapter (1...N)
+.IP "\fBplaylist\fR" 4
+.IX Item "playlist"
+Playlist to read (\s-1BDMV/PLAYLIST/\s0?????.mpls)
+.PP
+Examples:
+.PP
+Read longest playlist from BluRay mounted to /mnt/bluray:
+.PP
+.Vb 1
+\& bluray:/mnt/bluray
+.Ve
+.PP
+Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
+.PP
+.Vb 1
+\& \-playlist 4 \-angle 2 \-chapter 2 bluray:/mnt/bluray
+.Ve
+.SS "cache"
+.IX Subsection "cache"
+Caching wrapper for input stream.
+.PP
+Cache the input stream to temporary file. It brings seeking capability to live streams.
+.PP
+.Vb 1
+\& cache:<URL>
+.Ve
+.SS "concat"
+.IX Subsection "concat"
+Physical concatenation protocol.
+.PP
+Read and seek from many resources in sequence as if they were
+a unique resource.
+.PP
+A \s-1URL\s0 accepted by this protocol has the syntax:
+.PP
+.Vb 1
+\& concat:<URL1>|<URL2>|...|<URLN>
+.Ve
+.PP
+where \fI\s-1URL1\s0\fR, \fI\s-1URL2\s0\fR, ..., \fI\s-1URLN\s0\fR are the urls of the
+resource to be concatenated, each one possibly specifying a distinct
+protocol.
+.PP
+For example to read a sequence of files \fIsplit1.mpeg\fR,
+\&\fIsplit2.mpeg\fR, \fIsplit3.mpeg\fR with \fBffplay\fR use the
+command:
+.PP
+.Vb 1
+\& ffplay concat:split1.mpeg\e|split2.mpeg\e|split3.mpeg
+.Ve
+.PP
+Note that you may need to escape the character \*(L"|\*(R" which is special for
+many shells.
+.SS "crypto"
+.IX Subsection "crypto"
+AES-encrypted stream reading protocol.
+.PP
+The accepted options are:
+.IP "\fBkey\fR" 4
+.IX Item "key"
+Set the \s-1AES\s0 decryption key binary block from given hexadecimal representation.
+.IP "\fBiv\fR" 4
+.IX Item "iv"
+Set the \s-1AES\s0 decryption initialization vector binary block from given hexadecimal representation.
+.PP
+Accepted \s-1URL\s0 formats:
+.PP
+.Vb 2
+\& crypto:<URL>
+\& crypto+<URL>
+.Ve
+.SS "data"
+.IX Subsection "data"
+Data in-line in the \s-1URI\s0. See <\fBhttp://en.wikipedia.org/wiki/Data_URI_scheme\fR>.
+.PP
+For example, to convert a \s-1GIF\s0 file given inline with \fBffmpeg\fR:
+.PP
+.Vb 1
+\& ffmpeg \-i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
+.Ve
+.SS "file"
+.IX Subsection "file"
+File access protocol.
+.PP
+Read from or write to a file.
+.PP
+A file \s-1URL\s0 can have the form:
+.PP
+.Vb 1
+\& file:<filename>
+.Ve
+.PP
+where \fIfilename\fR is the path of the file to read.
+.PP
+An \s-1URL\s0 that does not have a protocol prefix will be assumed to be a
+file \s-1URL\s0. Depending on the build, an \s-1URL\s0 that looks like a Windows
+path with the drive letter at the beginning will also be assumed to be
+a file \s-1URL\s0 (usually not the case in builds for unix-like systems).
+.PP
+For example to read from a file \fIinput.mpeg\fR with \fBffmpeg\fR
+use the command:
+.PP
+.Vb 1
+\& ffmpeg \-i file:input.mpeg output.mpeg
+.Ve
+.PP
+This protocol accepts the following options:
+.IP "\fBtruncate\fR" 4
+.IX Item "truncate"
+Truncate existing files on write, if set to 1. A value of 0 prevents
+truncating. Default value is 1.
+.IP "\fBblocksize\fR" 4
+.IX Item "blocksize"
+Set I/O operation maximum block size, in bytes. Default value is
+\&\f(CW\*(C`INT_MAX\*(C'\fR, 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.
+.SS "ftp"
+.IX Subsection "ftp"
+\&\s-1FTP\s0 (File Transfer Protocol).
+.PP
+Read from or write to remote resources using \s-1FTP\s0 protocol.
+.PP
+Following syntax is required.
+.PP
+.Vb 1
+\& ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
+.Ve
+.PP
+This protocol accepts the following options.
+.IP "\fBtimeout\fR" 4
+.IX Item "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.
+.IP "\fBftp-anonymous-password\fR" 4
+.IX Item "ftp-anonymous-password"
+Password used when login as anonymous user. Typically an e\-mail address
+should be used.
+.IP "\fBftp-write-seekable\fR" 4
+.IX Item "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.
+.PP
+\&\s-1NOTE:\s0 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 \s-1FTP\s0 servers behave in different way during seek
+operation. ff* tools may produce incomplete content due to server limitations.
+.PP
+This protocol accepts the following options:
+.IP "\fBfollow\fR" 4
+.IX Item "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 \s-1API\s0 users).
+.SS "gopher"
+.IX Subsection "gopher"
+Gopher protocol.
+.SS "hls"
+.IX Subsection "hls"
+Read Apple \s-1HTTP\s0 Live Streaming compliant segmented stream as
+a uniform one. The M3U8 playlists describing the segments can be
+remote \s-1HTTP\s0 resources or local files, accessed using the standard
+file protocol.
+The nested protocol is declared by specifying
+"+\fIproto\fR" after the hls \s-1URI\s0 scheme name, where \fIproto\fR
+is either \*(L"file\*(R" or \*(L"http\*(R".
+.PP
+.Vb 2
+\& hls+http://host/path/to/remote/resource.m3u8
+\& hls+file://path/to/local/resource.m3u8
+.Ve
+.PP
+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.
+.SS "http"
+.IX Subsection "http"
+\&\s-1HTTP\s0 (Hyper Text Transfer Protocol).
+.PP
+This protocol accepts the following options:
+.IP "\fBseekable\fR" 4
+.IX 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.
+.IP "\fBchunked_post\fR" 4
+.IX Item "chunked_post"
+If set to 1 use chunked Transfer-Encoding for posts, default is 1.
+.IP "\fBcontent_type\fR" 4
+.IX Item "content_type"
+Set a specific content type for the \s-1POST\s0 messages or for listen mode.
+.IP "\fBhttp_proxy\fR" 4
+.IX Item "http_proxy"
+set \s-1HTTP\s0 proxy to tunnel through e.g. http://example.com:1234
+.IP "\fBheaders\fR" 4
+.IX Item "headers"
+Set custom \s-1HTTP\s0 headers, can override built in default headers. The
+value must be a string encoding the headers.
+.IP "\fBmultiple_requests\fR" 4
+.IX Item "multiple_requests"
+Use persistent connections if set to 1, default is 0.
+.IP "\fBpost_data\fR" 4
+.IX Item "post_data"
+Set custom \s-1HTTP\s0 post data.
+.IP "\fBreferer\fR" 4
+.IX Item "referer"
+Set the Referer header. Include 'Referer: \s-1URL\s0' header in \s-1HTTP\s0 request.
+.IP "\fBuser_agent\fR" 4
+.IX Item "user_agent"
+Override the User-Agent header. If not specified the protocol will use a
+string describing the libavformat build. (\*(L"Lavf/<version>\*(R")
+.IP "\fBuser-agent\fR" 4
+.IX Item "user-agent"
+This is a deprecated option, you can use user_agent instead it.
+.IP "\fBtimeout\fR" 4
+.IX Item "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.
+.IP "\fBreconnect_at_eof\fR" 4
+.IX Item "reconnect_at_eof"
+If set then eof is treated like an error and causes reconnection, this is useful
+for live / endless streams.
+.IP "\fBreconnect_streamed\fR" 4
+.IX Item "reconnect_streamed"
+If set then even streamed/non seekable streams will be reconnected on errors.
+.IP "\fBreconnect_delay_max\fR" 4
+.IX Item "reconnect_delay_max"
+Sets the maximum delay in seconds after which to give up reconnecting
+.IP "\fBmime_type\fR" 4
+.IX Item "mime_type"
+Export the \s-1MIME\s0 type.
+.IP "\fBhttp_version\fR" 4
+.IX Item "http_version"
+Exports the \s-1HTTP\s0 response version number. Usually \*(L"1.0\*(R" or \*(L"1.1\*(R".
+.IP "\fBicy\fR" 4
+.IX Item "icy"
+If set to 1 request \s-1ICY\s0 (SHOUTcast) metadata from the server. If the server
+supports this, the metadata has to be retrieved by the application by reading
+the \fBicy_metadata_headers\fR and \fBicy_metadata_packet\fR options.
+The default is 1.
+.IP "\fBicy_metadata_headers\fR" 4
+.IX Item "icy_metadata_headers"
+If the server supports \s-1ICY\s0 metadata, this contains the ICY-specific \s-1HTTP\s0 reply
+headers, separated by newline characters.
+.IP "\fBicy_metadata_packet\fR" 4
+.IX Item "icy_metadata_packet"
+If the server supports \s-1ICY\s0 metadata, and \fBicy\fR 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.
+.IP "\fBcookies\fR" 4
+.IX Item "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 \s-1HTTP\s0 response field. Multiple cookies can be
+delimited by a newline character.
+.IP "\fBoffset\fR" 4
+.IX Item "offset"
+Set initial byte offset.
+.IP "\fBend_offset\fR" 4
+.IX Item "end_offset"
+Try to limit the request to bytes preceding this offset.
+.IP "\fBmethod\fR" 4
+.IX Item "method"
+When used as a client option it sets the \s-1HTTP\s0 method for the request.
+.Sp
+When used as a server option it sets the \s-1HTTP\s0 method that is going to be
+expected from the client(s).
+If the expected and the received \s-1HTTP\s0 method do not match the client will
+be given a Bad Request response.
+When unset the \s-1HTTP\s0 method is not checked for now. This will be replaced by
+autodetection in the future.
+.IP "\fBlisten\fR" 4
+.IX Item "listen"
+If set to 1 enables experimental \s-1HTTP\s0 server. This can be used to send data when
+used as an output option, or read data from a client with \s-1HTTP\s0 \s-1POST\s0 when used as
+an input option.
+If set to 2 enables experimental multi-client \s-1HTTP\s0 server. This is not yet implemented
+in ffmpeg.c and thus must not be used as a command line option.
+.Sp
+.Vb 2
+\& # 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>
+.Ve
+.PP
+\fI\s-1HTTP\s0 Cookies\fR
+.IX Subsection "HTTP Cookies"
+.PP
+Some \s-1HTTP\s0 requests will be denied unless cookie values are passed in with the
+request. The \fBcookies\fR option allows these cookies to be specified. At
+the very least, each cookie must specify a value along with a path and domain.
+\&\s-1HTTP\s0 requests that match both the domain and path will automatically include the
+cookie value in the \s-1HTTP\s0 Cookie header field. Multiple cookies can be delimited
+by a newline.
+.PP
+The required syntax to play a stream specifying a cookie is:
+.PP
+.Vb 1
+\& ffplay \-cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
+.Ve
+.SS "Icecast"
+.IX Subsection "Icecast"
+Icecast protocol (stream to Icecast servers)
+.PP
+This protocol accepts the following options:
+.IP "\fBice_genre\fR" 4
+.IX Item "ice_genre"
+Set the stream genre.
+.IP "\fBice_name\fR" 4
+.IX Item "ice_name"
+Set the stream name.
+.IP "\fBice_description\fR" 4
+.IX Item "ice_description"
+Set the stream description.
+.IP "\fBice_url\fR" 4
+.IX Item "ice_url"
+Set the stream website \s-1URL\s0.
+.IP "\fBice_public\fR" 4
+.IX Item "ice_public"
+Set if the stream should be public.
+The default is 0 (not public).
+.IP "\fBuser_agent\fR" 4
+.IX Item "user_agent"
+Override the User-Agent header. If not specified a string of the form
+\&\*(L"Lavf/<version>\*(R" will be used.
+.IP "\fBpassword\fR" 4
+.IX Item "password"
+Set the Icecast mountpoint password.
+.IP "\fBcontent_type\fR" 4
+.IX Item "content_type"
+Set the stream content type. This must be set if it is different from
+audio/mpeg.
+.IP "\fBlegacy_icecast\fR" 4
+.IX Item "legacy_icecast"
+This enables support for Icecast versions < 2.4.0, that do not support the
+\&\s-1HTTP\s0 \s-1PUT\s0 method but the \s-1SOURCE\s0 method.
+.PP
+.Vb 1
+\& icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>
+.Ve
+.SS "mmst"
+.IX Subsection "mmst"
+\&\s-1MMS\s0 (Microsoft Media Server) protocol over \s-1TCP\s0.
+.SS "mmsh"
+.IX Subsection "mmsh"
+\&\s-1MMS\s0 (Microsoft Media Server) protocol over \s-1HTTP\s0.
+.PP
+The required syntax is:
+.PP
+.Vb 1
+\& mmsh://<server>[:<port>][/<app>][/<playpath>]
+.Ve
+.SS "md5"
+.IX Subsection "md5"
+\&\s-1MD5\s0 output protocol.
+.PP
+Computes the \s-1MD5\s0 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.
+.PP
+Some examples follow.
+.PP
+.Vb 2
+\& # 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:
+.Ve
+.PP
+Note that some formats (typically \s-1MOV\s0) require the output protocol to
+be seekable, so they will fail with the \s-1MD5\s0 output protocol.
+.SS "pipe"
+.IX Subsection "pipe"
+\&\s-1UNIX\s0 pipe access protocol.
+.PP
+Read and write from \s-1UNIX\s0 pipes.
+.PP
+The accepted syntax is:
+.PP
+.Vb 1
+\& pipe:[<number>]
+.Ve
+.PP
+\&\fInumber\fR is the number corresponding to the file descriptor of the
+pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If \fInumber\fR
+is not specified, by default the stdout file descriptor will be used
+for writing, stdin for reading.
+.PP
+For example to read from stdin with \fBffmpeg\fR:
+.PP
+.Vb 3
+\& cat test.wav | ffmpeg \-i pipe:0
+\& # ...this is the same as...
+\& cat test.wav | ffmpeg \-i pipe:
+.Ve
+.PP
+For writing to stdout with \fBffmpeg\fR:
+.PP
+.Vb 3
+\& 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
+.Ve
+.PP
+This protocol accepts the following options:
+.IP "\fBblocksize\fR" 4
+.IX Item "blocksize"
+Set I/O operation maximum block size, in bytes. Default value is
+\&\f(CW\*(C`INT_MAX\*(C'\fR, 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.
+.PP
+Note that some formats (typically \s-1MOV\s0), require the output protocol to
+be seekable, so they will fail with the pipe output protocol.
+.SS "prompeg"
+.IX Subsection "prompeg"
+Pro-MPEG Code of Practice #3 Release 2 \s-1FEC\s0 protocol.
+.PP
+The Pro-MPEG CoP#3 \s-1FEC\s0 is a 2D parity-check forward error correction mechanism
+for \s-1MPEG\-2\s0 Transport Streams sent over \s-1RTP\s0.
+.PP
+This protocol must be used in conjunction with the \f(CW\*(C`rtp_mpegts\*(C'\fR muxer and
+the \f(CW\*(C`rtp\*(C'\fR protocol.
+.PP
+The required syntax is:
+.PP
+.Vb 1
+\& \-f rtp_mpegts \-fec prompeg=<option>=<val>... rtp://<hostname>:<port>
+.Ve
+.PP
+The destination \s-1UDP\s0 ports are \f(CW\*(C`port + 2\*(C'\fR for the column \s-1FEC\s0 stream
+and \f(CW\*(C`port + 4\*(C'\fR for the row \s-1FEC\s0 stream.
+.PP
+This protocol accepts the following options:
+.IP "\fBl=\fR\fIn\fR" 4
+.IX Item "l=n"
+The number of columns (4\-20, LxD <= 100)
+.IP "\fBd=\fR\fIn\fR" 4
+.IX Item "d=n"
+The number of rows (4\-20, LxD <= 100)
+.PP
+Example usage:
+.PP
+.Vb 1
+\& \-f rtp_mpegts \-fec prompeg=l=8:d=4 rtp://<hostname>:<port>
+.Ve
+.SS "rtmp"
+.IX Subsection "rtmp"
+Real-Time Messaging Protocol.
+.PP
+The Real-Time Messaging Protocol (\s-1RTMP\s0) is used for streaming multimedia
+content across a \s-1TCP/IP\s0 network.
+.PP
+The required syntax is:
+.PP
+.Vb 1
+\& rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]
+.Ve
+.PP
+The accepted parameters are:
+.IP "\fBusername\fR" 4
+.IX Item "username"
+An optional username (mostly for publishing).
+.IP "\fBpassword\fR" 4
+.IX Item "password"
+An optional password (mostly for publishing).
+.IP "\fBserver\fR" 4
+.IX Item "server"
+The address of the \s-1RTMP\s0 server.
+.IP "\fBport\fR" 4
+.IX Item "port"
+The number of the \s-1TCP\s0 port to use (by default is 1935).
+.IP "\fBapp\fR" 4
+.IX Item "app"
+It is the name of the application to access. It usually corresponds to
+the path where the application is installed on the \s-1RTMP\s0 server
+(e.g. \fI/ondemand/\fR, \fI/flash/live/\fR, etc.). You can override
+the value parsed from the \s-1URI\s0 through the \f(CW\*(C`rtmp_app\*(C'\fR option, too.
+.IP "\fBplaypath\fR" 4
+.IX Item "playpath"
+It is the path or name of the resource to play with reference to the
+application specified in \fIapp\fR, may be prefixed by \*(L"mp4:\*(R". You
+can override the value parsed from the \s-1URI\s0 through the \f(CW\*(C`rtmp_playpath\*(C'\fR
+option, too.
+.IP "\fBlisten\fR" 4
+.IX Item "listen"
+Act as a server, listening for an incoming connection.
+.IP "\fBtimeout\fR" 4
+.IX Item "timeout"
+Maximum time to wait for the incoming connection. Implies listen.
+.PP
+Additionally, the following parameters can be set via command line options
+(or in code via \f(CW\*(C`AVOption\*(C'\fRs):
+.IP "\fBrtmp_app\fR" 4
+.IX Item "rtmp_app"
+Name of application to connect on the \s-1RTMP\s0 server. This option
+overrides the parameter specified in the \s-1URI\s0.
+.IP "\fBrtmp_buffer\fR" 4
+.IX Item "rtmp_buffer"
+Set the client buffer time in milliseconds. The default is 3000.
+.IP "\fBrtmp_conn\fR" 4
+.IX Item "rtmp_conn"
+Extra arbitrary \s-1AMF\s0 connection parameters, parsed from a string,
+e.g. like \f(CW\*(C`B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0\*(C'\fR.
+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
+\&\s-1FALSE\s0 or \s-1TRUE\s0, 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. \f(CW\*(C`NB:myFlag:1\*(C'\fR). This option may be used multiple
+times to construct arbitrary \s-1AMF\s0 sequences.
+.IP "\fBrtmp_flashver\fR" 4
+.IX Item "rtmp_flashver"
+Version of the Flash plugin used to run the \s-1SWF\s0 player. The default
+is \s-1LNX\s0 9,0,124,2. (When publishing, the default is \s-1FMLE/3\s0.0 (compatible;
+<libavformat version>).)
+.IP "\fBrtmp_flush_interval\fR" 4
+.IX Item "rtmp_flush_interval"
+Number of packets flushed in the same request (\s-1RTMPT\s0 only). The default
+is 10.
+.IP "\fBrtmp_live\fR" 4
+.IX Item "rtmp_live"
+Specify that the media is a live stream. No resuming or seeking in
+live streams is possible. The default value is \f(CW\*(C`any\*(C'\fR, 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 \f(CW\*(C`live\*(C'\fR and
+\&\f(CW\*(C`recorded\*(C'\fR.
+.IP "\fBrtmp_pageurl\fR" 4
+.IX Item "rtmp_pageurl"
+\&\s-1URL\s0 of the web page in which the media was embedded. By default no
+value will be sent.
+.IP "\fBrtmp_playpath\fR" 4
+.IX Item "rtmp_playpath"
+Stream identifier to play or to publish. This option overrides the
+parameter specified in the \s-1URI\s0.
+.IP "\fBrtmp_subscribe\fR" 4
+.IX 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.
+.IP "\fBrtmp_swfhash\fR" 4
+.IX Item "rtmp_swfhash"
+\&\s-1SHA256\s0 hash of the decompressed \s-1SWF\s0 file (32 bytes).
+.IP "\fBrtmp_swfsize\fR" 4
+.IX Item "rtmp_swfsize"
+Size of the decompressed \s-1SWF\s0 file, required for SWFVerification.
+.IP "\fBrtmp_swfurl\fR" 4
+.IX Item "rtmp_swfurl"
+\&\s-1URL\s0 of the \s-1SWF\s0 player for the media. By default no value will be sent.
+.IP "\fBrtmp_swfverify\fR" 4
+.IX Item "rtmp_swfverify"
+\&\s-1URL\s0 to player swf file, compute hash/size automatically.
+.IP "\fBrtmp_tcurl\fR" 4
+.IX Item "rtmp_tcurl"
+\&\s-1URL\s0 of the target stream. Defaults to proto://host[:port]/app.
+.PP
+For example to read with \fBffplay\fR a multimedia resource named
+\&\*(L"sample\*(R" from the application \*(L"vod\*(R" from an \s-1RTMP\s0 server \*(L"myserver\*(R":
+.PP
+.Vb 1
+\& ffplay rtmp://myserver/vod/sample
+.Ve
+.PP
+To publish to a password protected server, passing the playpath and
+app names separately:
+.PP
+.Vb 1
+\& ffmpeg \-re \-i <input> \-f flv \-rtmp_playpath some/long/path \-rtmp_app long/app/name rtmp://username:password@myserver/
+.Ve
+.SS "rtmpe"
+.IX Subsection "rtmpe"
+Encrypted Real-Time Messaging Protocol.
+.PP
+The Encrypted Real-Time Messaging Protocol (\s-1RTMPE\s0) is used for
+streaming multimedia content within standard cryptographic primitives,
+consisting of Diffie-Hellman key exchange and \s-1HMACSHA256\s0, generating
+a pair of \s-1RC4\s0 keys.
+.SS "rtmps"
+.IX Subsection "rtmps"
+Real-Time Messaging Protocol over a secure \s-1SSL\s0 connection.
+.PP
+The Real-Time Messaging Protocol (\s-1RTMPS\s0) is used for streaming
+multimedia content across an encrypted connection.
+.SS "rtmpt"
+.IX Subsection "rtmpt"
+Real-Time Messaging Protocol tunneled through \s-1HTTP\s0.
+.PP
+The Real-Time Messaging Protocol tunneled through \s-1HTTP\s0 (\s-1RTMPT\s0) is used
+for streaming multimedia content within \s-1HTTP\s0 requests to traverse
+firewalls.
+.SS "rtmpte"
+.IX Subsection "rtmpte"
+Encrypted Real-Time Messaging Protocol tunneled through \s-1HTTP\s0.
+.PP
+The Encrypted Real-Time Messaging Protocol tunneled through \s-1HTTP\s0 (\s-1RTMPTE\s0)
+is used for streaming multimedia content within \s-1HTTP\s0 requests to traverse
+firewalls.
+.SS "rtmpts"
+.IX Subsection "rtmpts"
+Real-Time Messaging Protocol tunneled through \s-1HTTPS\s0.
+.PP
+The Real-Time Messaging Protocol tunneled through \s-1HTTPS\s0 (\s-1RTMPTS\s0) is used
+for streaming multimedia content within \s-1HTTPS\s0 requests to traverse
+firewalls.
+.SS "libsmbclient"
+.IX Subsection "libsmbclient"
+libsmbclient permits one to manipulate \s-1CIFS/SMB\s0 network resources.
+.PP
+Following syntax is required.
+.PP
+.Vb 1
+\& smb://[[domain:]user[:password@]]server[/share[/path[/file]]]
+.Ve
+.PP
+This protocol accepts the following options.
+.IP "\fBtimeout\fR" 4
+.IX Item "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.
+.IP "\fBtruncate\fR" 4
+.IX Item "truncate"
+Truncate existing files on write, if set to 1. A value of 0 prevents
+truncating. Default value is 1.
+.IP "\fBworkgroup\fR" 4
+.IX Item "workgroup"
+Set the workgroup used for making connections. By default workgroup is not specified.
+.PP
+For more information see: <\fBhttp://www.samba.org/\fR>.
+.SS "libssh"
+.IX Subsection "libssh"
+Secure File Transfer Protocol via libssh
+.PP
+Read from or write to remote resources using \s-1SFTP\s0 protocol.
+.PP
+Following syntax is required.
+.PP
+.Vb 1
+\& sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
+.Ve
+.PP
+This protocol accepts the following options.
+.IP "\fBtimeout\fR" 4
+.IX 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.
+.IP "\fBtruncate\fR" 4
+.IX Item "truncate"
+Truncate existing files on write, if set to 1. A value of 0 prevents
+truncating. Default value is 1.
+.IP "\fBprivate_key\fR" 4
+.IX Item "private_key"
+Specify the path of the file containing private key to use during authorization.
+By default libssh searches for keys in the \fI~/.ssh/\fR directory.
+.PP
+Example: Play a file stored on remote server.
+.PP
+.Vb 1
+\& ffplay sftp://user:password@server_address:22/home/user/resource.mpeg
+.Ve
+.SS "librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte"
+.IX Subsection "librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte"
+Real-Time Messaging Protocol and its variants supported through
+librtmp.
+.PP
+Requires the presence of the librtmp headers and library during
+configuration. You need to explicitly configure the build with
+\&\*(L"\-\-enable\-librtmp\*(R". If enabled this will replace the native \s-1RTMP\s0
+protocol.
+.PP
+This protocol provides most client functions and a few server
+functions needed to support \s-1RTMP\s0, \s-1RTMP\s0 tunneled in \s-1HTTP\s0 (\s-1RTMPT\s0),
+encrypted \s-1RTMP\s0 (\s-1RTMPE\s0), \s-1RTMP\s0 over \s-1SSL/TLS\s0 (\s-1RTMPS\s0) and tunneled
+variants of these encrypted types (\s-1RTMPTE\s0, \s-1RTMPTS\s0).
+.PP
+The required syntax is:
+.PP
+.Vb 1
+\& <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>
+.Ve
+.PP
+where \fIrtmp_proto\fR is one of the strings \*(L"rtmp\*(R", \*(L"rtmpt\*(R", \*(L"rtmpe\*(R",
+\&\*(L"rtmps\*(R", \*(L"rtmpte\*(R", \*(L"rtmpts\*(R" corresponding to each \s-1RTMP\s0 variant, and
+\&\fIserver\fR, \fIport\fR, \fIapp\fR and \fIplaypath\fR have the same
+meaning as specified for the \s-1RTMP\s0 native protocol.
+\&\fIoptions\fR contains a list of space-separated options of the form
+\&\fIkey\fR=\fIval\fR.
+.PP
+See the librtmp manual page (man 3 librtmp) for more information.
+.PP
+For example, to stream a file in real-time to an \s-1RTMP\s0 server using
+\&\fBffmpeg\fR:
+.PP
+.Vb 1
+\& ffmpeg \-re \-i myfile \-f flv rtmp://myserver/live/mystream
+.Ve
+.PP
+To play the same stream using \fBffplay\fR:
+.PP
+.Vb 1
+\& ffplay "rtmp://myserver/live/mystream live=1"
+.Ve
+.SS "rtp"
+.IX Subsection "rtp"
+Real-time Transport Protocol.
+.PP
+The required syntax for an \s-1RTP\s0 \s-1URL\s0 is:
+rtp://\fIhostname\fR[:\fIport\fR][?\fIoption\fR=\fIval\fR...]
+.PP
+\&\fIport\fR specifies the \s-1RTP\s0 port to use.
+.PP
+The following \s-1URL\s0 options are supported:
+.IP "\fBttl=\fR\fIn\fR" 4
+.IX Item "ttl=n"
+Set the \s-1TTL\s0 (Time-To-Live) value (for multicast only).
+.IP "\fBrtcpport=\fR\fIn\fR" 4
+.IX Item "rtcpport=n"
+Set the remote \s-1RTCP\s0 port to \fIn\fR.
+.IP "\fBlocalrtpport=\fR\fIn\fR" 4
+.IX Item "localrtpport=n"
+Set the local \s-1RTP\s0 port to \fIn\fR.
+.IP "\fBlocalrtcpport=\fR\fIn\fR\fB'\fR" 4
+.IX Item "localrtcpport=n'"
+Set the local \s-1RTCP\s0 port to \fIn\fR.
+.IP "\fBpkt_size=\fR\fIn\fR" 4
+.IX Item "pkt_size=n"
+Set max packet size (in bytes) to \fIn\fR.
+.IP "\fBconnect=0|1\fR" 4
+.IX Item "connect=0|1"
+Do a \f(CW\*(C`connect()\*(C'\fR on the \s-1UDP\s0 socket (if set to 1) or not (if set
+to 0).
+.IP "\fBsources=\fR\fIip\fR\fB[,\fR\fIip\fR\fB]\fR" 4
+.IX Item "sources=ip[,ip]"
+List allowed source \s-1IP\s0 addresses.
+.IP "\fBblock=\fR\fIip\fR\fB[,\fR\fIip\fR\fB]\fR" 4
+.IX Item "block=ip[,ip]"
+List disallowed (blocked) source \s-1IP\s0 addresses.
+.IP "\fBwrite_to_source=0|1\fR" 4
+.IX Item "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).
+.IP "\fBlocalport=\fR\fIn\fR" 4
+.IX Item "localport=n"
+Set the local \s-1RTP\s0 port to \fIn\fR.
+.Sp
+This is a deprecated option. Instead, \fBlocalrtpport\fR should be
+used.
+.PP
+Important notes:
+.IP "1." 4
+If \fBrtcpport\fR is not set the \s-1RTCP\s0 port will be set to the \s-1RTP\s0
+port value plus 1.
+.IP "2." 4
+If \fBlocalrtpport\fR (the local \s-1RTP\s0 port) is not set any available
+port will be used for the local \s-1RTP\s0 and \s-1RTCP\s0 ports.
+.IP "3." 4
+If \fBlocalrtcpport\fR (the local \s-1RTCP\s0 port) is not set it will be
+set to the local \s-1RTP\s0 port value plus 1.
+.SS "rtsp"
+.IX Subsection "rtsp"
+Real-Time Streaming Protocol.
+.PP
+\&\s-1RTSP\s0 is not technically a protocol handler in libavformat, it is a demuxer
+and muxer. The demuxer supports both normal \s-1RTSP\s0 (with data transferred
+over \s-1RTP\s0; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
+data transferred over \s-1RDT\s0).
+.PP
+The muxer can be used to send a stream using \s-1RTSP\s0 \s-1ANNOUNCE\s0 to a server
+supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
+<\fBhttps://github.com/revmischa/rtsp\-server\fR>).
+.PP
+The required syntax for a \s-1RTSP\s0 url is:
+.PP
+.Vb 1
+\& rtsp://<hostname>[:<port>]/<path>
+.Ve
+.PP
+Options can be set on the \fBffmpeg\fR/\fBffplay\fR command
+line, or set in code via \f(CW\*(C`AVOption\*(C'\fRs or in
+\&\f(CW\*(C`avformat_open_input\*(C'\fR.
+.PP
+The following options are supported.
+.IP "\fBinitial_pause\fR" 4
+.IX Item "initial_pause"
+Do not start playing the stream immediately if set to 1. Default value
+is 0.
+.IP "\fBrtsp_transport\fR" 4
+.IX Item "rtsp_transport"
+Set \s-1RTSP\s0 transport protocols.
+.Sp
+It accepts the following values:
+.RS 4
+.IP "\fBudp\fR" 4
+.IX Item "udp"
+Use \s-1UDP\s0 as lower transport protocol.
+.IP "\fBtcp\fR" 4
+.IX Item "tcp"
+Use \s-1TCP\s0 (interleaving within the \s-1RTSP\s0 control channel) as lower
+transport protocol.
+.IP "\fBudp_multicast\fR" 4
+.IX Item "udp_multicast"
+Use \s-1UDP\s0 multicast as lower transport protocol.
+.IP "\fBhttp\fR" 4
+.IX Item "http"
+Use \s-1HTTP\s0 tunneling as lower transport protocol, which is useful for
+passing proxies.
+.RE
+.RS 4
+.Sp
+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 \fBtcp\fR and \fBudp\fR options are supported.
+.RE
+.IP "\fBrtsp_flags\fR" 4
+.IX Item "rtsp_flags"
+Set \s-1RTSP\s0 flags.
+.Sp
+The following values are accepted:
+.RS 4
+.IP "\fBfilter_src\fR" 4
+.IX Item "filter_src"
+Accept packets only from negotiated peer address and port.
+.IP "\fBlisten\fR" 4
+.IX Item "listen"
+Act as a server, listening for an incoming connection.
+.IP "\fBprefer_tcp\fR" 4
+.IX Item "prefer_tcp"
+Try \s-1TCP\s0 for \s-1RTP\s0 transport first, if \s-1TCP\s0 is available as \s-1RTSP\s0 \s-1RTP\s0 transport.
+.RE
+.RS 4
+.Sp
+Default value is \fBnone\fR.
+.RE
+.IP "\fBallowed_media_types\fR" 4
+.IX Item "allowed_media_types"
+Set media types to accept from the server.
+.Sp
+The following flags are accepted:
+.RS 4
+.IP "\fBvideo\fR" 4
+.IX Item "video"
+.PD 0
+.IP "\fBaudio\fR" 4
+.IX Item "audio"
+.IP "\fBdata\fR" 4
+.IX Item "data"
+.RE
+.RS 4
+.PD
+.Sp
+By default it accepts all media types.
+.RE
+.IP "\fBmin_port\fR" 4
+.IX Item "min_port"
+Set minimum local \s-1UDP\s0 port. Default value is 5000.
+.IP "\fBmax_port\fR" 4
+.IX Item "max_port"
+Set maximum local \s-1UDP\s0 port. Default value is 65000.
+.IP "\fBtimeout\fR" 4
+.IX Item "timeout"
+Set maximum timeout (in seconds) to wait for incoming connections.
+.Sp
+A value of \-1 means infinite (default). This option implies the
+\&\fBrtsp_flags\fR set to \fBlisten\fR.
+.IP "\fBreorder_queue_size\fR" 4
+.IX Item "reorder_queue_size"
+Set number of packets to buffer for handling of reordered packets.
+.IP "\fBstimeout\fR" 4
+.IX Item "stimeout"
+Set socket \s-1TCP\s0 I/O timeout in microseconds.
+.IP "\fBuser-agent\fR" 4
+.IX Item "user-agent"
+Override User-Agent header. If not specified, it defaults to the
+libavformat identifier string.
+.PP
+When receiving data over \s-1UDP\s0, 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 \f(CW\*(C`max_delay\*(C'\fR field of AVFormatContext).
+.PP
+When watching multi-bitrate Real-RTSP streams with \fBffplay\fR, the
+streams to display can be chosen with \f(CW\*(C`\-vst\*(C'\fR \fIn\fR and
+\&\f(CW\*(C`\-ast\*(C'\fR \fIn\fR for video and audio respectively, and can be switched
+on the fly by pressing \f(CW\*(C`v\*(C'\fR and \f(CW\*(C`a\*(C'\fR.
+.PP
+\fIExamples\fR
+.IX Subsection "Examples"
+.PP
+The following examples all make use of the \fBffplay\fR and
+\&\fBffmpeg\fR tools.
+.IP "\(bu" 4
+Watch a stream over \s-1UDP\s0, with a max reordering delay of 0.5 seconds:
+.Sp
+.Vb 1
+\& ffplay \-max_delay 500000 \-rtsp_transport udp rtsp://server/video.mp4
+.Ve
+.IP "\(bu" 4
+Watch a stream tunneled over \s-1HTTP:\s0
+.Sp
+.Vb 1
+\& ffplay \-rtsp_transport http rtsp://server/video.mp4
+.Ve
+.IP "\(bu" 4
+Send a stream in realtime to a \s-1RTSP\s0 server, for others to watch:
+.Sp
+.Vb 1
+\& ffmpeg \-re \-i <input> \-f rtsp \-muxdelay 0.1 rtsp://server/live.sdp
+.Ve
+.IP "\(bu" 4
+Receive a stream in realtime:
+.Sp
+.Vb 1
+\& ffmpeg \-rtsp_flags listen \-i rtsp://ownaddress/live.sdp <output>
+.Ve
+.SS "sap"
+.IX Subsection "sap"
+Session Announcement Protocol (\s-1RFC\s0 2974). This is not technically a
+protocol handler in libavformat, it is a muxer and demuxer.
+It is used for signalling of \s-1RTP\s0 streams, by announcing the \s-1SDP\s0 for the
+streams regularly on a separate port.
+.PP
+\fIMuxer\fR
+.IX Subsection "Muxer"
+.PP
+The syntax for a \s-1SAP\s0 url given to the muxer is:
+.PP
+.Vb 1
+\& sap://<destination>[:<port>][?<options>]
+.Ve
+.PP
+The \s-1RTP\s0 packets are sent to \fIdestination\fR on port \fIport\fR,
+or to port 5004 if no port is specified.
+\&\fIoptions\fR is a \f(CW\*(C`&\*(C'\fR\-separated list. The following options
+are supported:
+.IP "\fBannounce_addr=\fR\fIaddress\fR" 4
+.IX Item "announce_addr=address"
+Specify the destination \s-1IP\s0 address for sending the announcements to.
+If omitted, the announcements are sent to the commonly used \s-1SAP\s0
+announcement multicast address 224.2.127.254 (sap.mcast.net), or
+ff0e::2:7ffe if \fIdestination\fR is an IPv6 address.
+.IP "\fBannounce_port=\fR\fIport\fR" 4
+.IX Item "announce_port=port"
+Specify the port to send the announcements on, defaults to
+9875 if not specified.
+.IP "\fBttl=\fR\fIttl\fR" 4
+.IX Item "ttl=ttl"
+Specify the time to live value for the announcements and \s-1RTP\s0 packets,
+defaults to 255.
+.IP "\fBsame_port=\fR\fI0|1\fR" 4
+.IX Item "same_port=0|1"
+If set to 1, send all \s-1RTP\s0 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 \s-1RTP\s0 stack in libavformat for receiving requires all streams to be sent
+on unique ports.
+.PP
+Example command lines follow.
+.PP
+To broadcast a stream on the local subnet, for watching in \s-1VLC:\s0
+.PP
+.Vb 1
+\& ffmpeg \-re \-i <input> \-f sap sap://224.0.0.255?same_port=1
+.Ve
+.PP
+Similarly, for watching in \fBffplay\fR:
+.PP
+.Vb 1
+\& ffmpeg \-re \-i <input> \-f sap sap://224.0.0.255
+.Ve
+.PP
+And for watching in \fBffplay\fR, over IPv6:
+.PP
+.Vb 1
+\& ffmpeg \-re \-i <input> \-f sap sap://[ff0e::1:2:3:4]
+.Ve
+.PP
+\fIDemuxer\fR
+.IX Subsection "Demuxer"
+.PP
+The syntax for a \s-1SAP\s0 url given to the demuxer is:
+.PP
+.Vb 1
+\& sap://[<address>][:<port>]
+.Ve
+.PP
+\&\fIaddress\fR is the multicast address to listen for announcements on,
+if omitted, the default 224.2.127.254 (sap.mcast.net) is used. \fIport\fR
+is the port that is listened on, 9875 if omitted.
+.PP
+The demuxers listens for announcements on the given address and port.
+Once an announcement is received, it tries to receive that particular stream.
+.PP
+Example command lines follow.
+.PP
+To play back the first stream announced on the normal \s-1SAP\s0 multicast address:
+.PP
+.Vb 1
+\& ffplay sap://
+.Ve
+.PP
+To play back the first stream announced on one the default IPv6 \s-1SAP\s0 multicast address:
+.PP
+.Vb 1
+\& ffplay sap://[ff0e::2:7ffe]
+.Ve
+.SS "sctp"
+.IX Subsection "sctp"
+Stream Control Transmission Protocol.
+.PP
+The accepted \s-1URL\s0 syntax is:
+.PP
+.Vb 1
+\& sctp://<host>:<port>[?<options>]
+.Ve
+.PP
+The protocol accepts the following options:
+.IP "\fBlisten\fR" 4
+.IX Item "listen"
+If set to any value, listen for an incoming connection. Outgoing connection is done by default.
+.IP "\fBmax_streams\fR" 4
+.IX Item "max_streams"
+Set the maximum number of streams. By default no limit is set.
+.SS "srt"
+.IX Subsection "srt"
+Haivision Secure Reliable Transport Protocol via libsrt.
+.PP
+The supported syntax for a \s-1SRT\s0 \s-1URL\s0 is:
+.PP
+.Vb 1
+\& srt://<hostname>:<port>[?<options>]
+.Ve
+.PP
+\&\fIoptions\fR contains a list of &\-separated options of the form
+\&\fIkey\fR=\fIval\fR.
+.PP
+or
+.PP
+.Vb 1
+\& <options> srt://<hostname>:<port>
+.Ve
+.PP
+\&\fIoptions\fR contains a list of '\-\fIkey\fR \fIval\fR'
+options.
+.PP
+This protocol accepts the following options.
+.IP "\fBconnect_timeout\fR" 4
+.IX Item "connect_timeout"
+Connection timeout; \s-1SRT\s0 cannot connect for \s-1RTT\s0 > 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).
+.IP "\fBffs=\fR\fIbytes\fR" 4
+.IX Item "ffs=bytes"
+Flight Flag Size (Window Size), in bytes. \s-1FFS\s0 is actually an
+internal parameter and you should set it to not less than
+\&\fBrecv_buffer_size\fR and \fBmss\fR. 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.
+.IP "\fBinputbw=\fR\fIbytes/seconds\fR" 4
+.IX Item "inputbw=bytes/seconds"
+Sender nominal input rate, in bytes per seconds. Used along with
+\&\fBoheadbw\fR, when \fBmaxbw\fR is set to relative (0), to
+calculate maximum sending rate when recovery packets are sent
+along with the main media stream:
+\&\fBinputbw\fR * (100 + \fBoheadbw\fR) / 100
+if \fBinputbw\fR is not set while \fBmaxbw\fR is set to
+relative (0), the actual input rate is evaluated inside
+the library. Default value is 0.
+.IP "\fBiptos=\fR\fItos\fR" 4
+.IX Item "iptos=tos"
+\&\s-1IP\s0 Type of Service. Applies to sender only. Default value is 0xB8.
+.IP "\fBipttl=\fR\fIttl\fR" 4
+.IX Item "ipttl=ttl"
+\&\s-1IP\s0 Time To Live. Applies to sender only. Default value is 64.
+.IP "\fBlatency\fR" 4
+.IX Item "latency"
+Timestamp-based Packet Delivery Delay.
+Used to absorb bursts of missed packet retransmissions.
+This flag sets both \fBrcvlatency\fR and \fBpeerlatency\fR
+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 \fBpeerlatency\fR,
+when side is sender and \fBrcvlatency\fR
+when side is receiver, and the bidirectional stream
+sending is not supported.
+.IP "\fBlisten_timeout\fR" 4
+.IX Item "listen_timeout"
+Set socket listen timeout.
+.IP "\fBmaxbw=\fR\fIbytes/seconds\fR" 4
+.IX Item "maxbw=bytes/seconds"
+Maximum sending bandwidth, in bytes per seconds.
+\&\-1 infinite (\s-1CSRTCC\s0 limit is 30mbps)
+0 relative to input rate (see \fBinputbw\fR)
+>0 absolute limit value
+Default value is 0 (relative)
+.IP "\fBmode=\fR\fIcaller|listener|rendezvous\fR" 4
+.IX Item "mode=caller|listener|rendezvous"
+Connection mode.
+\&\fBcaller\fR opens client connection.
+\&\fBlistener\fR starts server to listen for incoming connections.
+\&\fBrendezvous\fR use Rendez-Vous connection mode.
+Default value is caller.
+.IP "\fBmss=\fR\fIbytes\fR" 4
+.IX Item "mss=bytes"
+Maximum Segment Size, in bytes. Used for buffer allocation
+and rate calculation using a packet counter assuming fully
+filled packets. The smallest \s-1MSS\s0 between the peers is
+used. This is 1500 by default in the overall internet.
+This is the maximum size of the \s-1UDP\s0 packet and can be
+only decreased, unless you have some unusual dedicated
+network settings. Default value is 1500.
+.IP "\fBnakreport=\fR\fI1|0\fR" 4
+.IX Item "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.
+.IP "\fBoheadbw=\fR\fIpercents\fR" 4
+.IX Item "oheadbw=percents"
+Recovery bandwidth overhead above input rate, in percents.
+See \fBinputbw\fR. Default value is 25%.
+.IP "\fBpassphrase=\fR\fIstring\fR" 4
+.IX Item "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 \s-1PBKDF2\s0
+(Password-Based Key Derivation Function). It is used
+only if \fBpbkeylen\fR is non-zero. It is used on
+the receiver only if the received data is encrypted.
+The configured passphrase cannot be recovered (write-only).
+.IP "\fBpayload_size=\fR\fIbytes\fR" 4
+.IX Item "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 \s-1SRT\s0
+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.
+.IP "\fBpkt_size=\fR\fIbytes\fR" 4
+.IX Item "pkt_size=bytes"
+Alias for \fBpayload_size\fR.
+.IP "\fBpeerlatency\fR" 4
+.IX Item "peerlatency"
+The latency value (as described in \fBrcvlatency\fR) that is
+set by the sender side as a minimum value for the receiver.
+.IP "\fBpbkeylen=\fR\fIbytes\fR" 4
+.IX Item "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.
+.IP "\fBrcvlatency\fR" 4
+.IX Item "rcvlatency"
+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 \s-1RTT\s0
+time, and the time needed to retransmit the lost \s-1UDP\s0
+packet. The effective latency value will be the maximum
+of this options' value and the value of \fBpeerlatency\fR
+set by the peer side. Before version 1.3.0 this option
+is only available as \fBlatency\fR.
+.IP "\fBrecv_buffer_size=\fR\fIbytes\fR" 4
+.IX Item "recv_buffer_size=bytes"
+Set \s-1UDP\s0 receive buffer size, expressed in bytes.
+.IP "\fBsend_buffer_size=\fR\fIbytes\fR" 4
+.IX Item "send_buffer_size=bytes"
+Set \s-1UDP\s0 send buffer size, expressed in bytes.
+.IP "\fBrw_timeout\fR" 4
+.IX Item "rw_timeout"
+Set raise error timeout for read/write optations.
+.Sp
+This option is only relevant in read mode:
+if no data arrived in more than this time
+interval, raise error.
+.IP "\fBtlpktdrop=\fR\fI1|0\fR" 4
+.IX Item "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 \s-1ACK\s0 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.
+.IP "\fBsndbuf=\fR\fIbytes\fR" 4
+.IX Item "sndbuf=bytes"
+Set send buffer size, expressed in bytes.
+.IP "\fBrcvbuf=\fR\fIbytes\fR" 4
+.IX Item "rcvbuf=bytes"
+Set receive buffer size, expressed in bytes.
+.Sp
+Receive buffer must not be greater than \fBffs\fR.
+.IP "\fBlossmaxttl=\fR\fIpackets\fR" 4
+.IX Item "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 \*(L"belated\*(R" packet has come, but it
+wasn't due to retransmission (that is, when \s-1UDP\s0 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 \*(L"gap\*(R" in sequences.
+.IP "\fBminversion\fR" 4
+.IX Item "minversion"
+The minimum \s-1SRT\s0 version that is required from the peer. A connection
+to a peer that does not satisfy the minimum version requirement
+will be rejected.
+.Sp
+The version format in hex is 0xXXYYZZ for x.y.z in human readable
+form.
+.IP "\fBstreamid=\fR\fIstring\fR" 4
+.IX Item "streamid=string"
+A string limited to 512 characters that can be set on the socket prior
+to connecting. This stream \s-1ID\s0 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 \s-1ID\s0. \s-1SRT\s0 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
+.IP "\fBsmoother=\fR\fIlive|file\fR" 4
+.IX Item "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.
+.IP "\fBmessageapi=\fR\fI1|0\fR" 4
+.IX Item "messageapi=1|0"
+When set, this socket uses the Message \s-1API\s0, otherwise it uses Buffer
+\&\s-1API\s0. Note that in live mode (see \fBtranstype\fR) thereXs only
+message \s-1API\s0 available. In File mode you can chose to use one of two modes:
+.Sp
+Stream \s-1API\s0 (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.
+.Sp
+Message \s-1API\s0. 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 \s-1UDP\s0 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.
+.IP "\fBtranstype=\fR\fIlive|file\fR" 4
+.IX Item "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.
+.Sp
+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 \s-1UDP\s0 packet,
+and limited to the value defined first in \fBpayload_size\fR (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).
+.Sp
+file: Set options as for non-live transmission. See \fBmessageapi\fR
+for further explanations
+.PP
+For more information see: <\fBhttps://github.com/Haivision/srt\fR>.
+.SS "srtp"
+.IX Subsection "srtp"
+Secure Real-time Transport Protocol.
+.PP
+The accepted options are:
+.IP "\fBsrtp_in_suite\fR" 4
+.IX Item "srtp_in_suite"
+.PD 0
+.IP "\fBsrtp_out_suite\fR" 4
+.IX Item "srtp_out_suite"
+.PD
+Select input and output encoding suites.
+.Sp
+Supported values:
+.RS 4
+.IP "\fB\s-1AES_CM_128_HMAC_SHA1_80\s0\fR" 4
+.IX Item "AES_CM_128_HMAC_SHA1_80"
+.PD 0
+.IP "\fB\s-1SRTP_AES128_CM_HMAC_SHA1_80\s0\fR" 4
+.IX Item "SRTP_AES128_CM_HMAC_SHA1_80"
+.IP "\fB\s-1AES_CM_128_HMAC_SHA1_32\s0\fR" 4
+.IX Item "AES_CM_128_HMAC_SHA1_32"
+.IP "\fB\s-1SRTP_AES128_CM_HMAC_SHA1_32\s0\fR" 4
+.IX Item "SRTP_AES128_CM_HMAC_SHA1_32"
+.RE
+.RS 4
+.RE
+.IP "\fBsrtp_in_params\fR" 4
+.IX Item "srtp_in_params"
+.IP "\fBsrtp_out_params\fR" 4
+.IX Item "srtp_out_params"
+.PD
+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.
+.SS "subfile"
+.IX Subsection "subfile"
+Virtually extract a segment of a file or another stream.
+The underlying stream must be seekable.
+.PP
+Accepted options:
+.IP "\fBstart\fR" 4
+.IX Item "start"
+Start offset of the extracted segment, in bytes.
+.IP "\fBend\fR" 4
+.IX Item "end"
+End offset of the extracted segment, in bytes.
+If set to 0, extract till end of file.
+.PP
+Examples:
+.PP
+Extract a chapter from a \s-1DVD\s0 \s-1VOB\s0 file (start and end sectors obtained
+externally and multiplied by 2048):
+.PP
+.Vb 1
+\& subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
+.Ve
+.PP
+Play an \s-1AVI\s0 file directly from a \s-1TAR\s0 archive:
+.PP
+.Vb 1
+\& subfile,,start,183241728,end,366490624,,:archive.tar
+.Ve
+.PP
+Play a MPEG-TS file from start offset till end:
+.PP
+.Vb 1
+\& subfile,,start,32815239,end,0,,:video.ts
+.Ve
+.SS "tee"
+.IX Subsection "tee"
+Writes the output to multiple protocols. The individual outputs are separated
+by |
+.PP
+.Vb 1
+\& tee:file://path/to/local/this.avi|file://path/to/local/that.avi
+.Ve
+.SS "tcp"
+.IX Subsection "tcp"
+Transmission Control Protocol.
+.PP
+The required syntax for a \s-1TCP\s0 url is:
+.PP
+.Vb 1
+\& tcp://<hostname>:<port>[?<options>]
+.Ve
+.PP
+\&\fIoptions\fR contains a list of &\-separated options of the form
+\&\fIkey\fR=\fIval\fR.
+.PP
+The list of supported options follows.
+.IP "\fBlisten=\fR\fI1|0\fR" 4
+.IX Item "listen=1|0"
+Listen for an incoming connection. Default value is 0.
+.IP "\fBtimeout=\fR\fImicroseconds\fR" 4
+.IX Item "timeout=microseconds"
+Set raise error timeout, expressed in microseconds.
+.Sp
+This option is only relevant in read mode: if no data arrived in more
+than this time interval, raise error.
+.IP "\fBlisten_timeout=\fR\fImilliseconds\fR" 4
+.IX Item "listen_timeout=milliseconds"
+Set listen timeout, expressed in milliseconds.
+.IP "\fBrecv_buffer_size=\fR\fIbytes\fR" 4
+.IX Item "recv_buffer_size=bytes"
+Set receive buffer size, expressed bytes.
+.IP "\fBsend_buffer_size=\fR\fIbytes\fR" 4
+.IX Item "send_buffer_size=bytes"
+Set send buffer size, expressed bytes.
+.IP "\fBtcp_nodelay=\fR\fI1|0\fR" 4
+.IX Item "tcp_nodelay=1|0"
+Set \s-1TCP_NODELAY\s0 to disable Nagle's algorithm. Default value is 0.
+.IP "\fBtcp_mss=\fR\fIbytes\fR" 4
+.IX Item "tcp_mss=bytes"
+Set maximum segment size for outgoing \s-1TCP\s0 packets, expressed in bytes.
+.PP
+The following example shows how to setup a listening \s-1TCP\s0 connection
+with \fBffmpeg\fR, which is then accessed with \fBffplay\fR:
+.PP
+.Vb 2
+\& ffmpeg \-i <input> \-f <format> tcp://<hostname>:<port>?listen
+\& ffplay tcp://<hostname>:<port>
+.Ve
+.SS "tls"
+.IX Subsection "tls"
+Transport Layer Security (\s-1TLS\s0) / Secure Sockets Layer (\s-1SSL\s0)
+.PP
+The required syntax for a \s-1TLS/SSL\s0 url is:
+.PP
+.Vb 1
+\& tls://<hostname>:<port>[?<options>]
+.Ve
+.PP
+The following parameters can be set via command line options
+(or in code via \f(CW\*(C`AVOption\*(C'\fRs):
+.IP "\fBca_file, cafile=\fR\fIfilename\fR" 4
+.IX Item "ca_file, cafile=filename"
+A file containing certificate authority (\s-1CA\s0) root certificates to treat
+as trusted. If the linked \s-1TLS\s0 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 \s-1PEM\s0 format.
+.IP "\fBtls_verify=\fR\fI1|0\fR" 4
+.IX Item "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 \s-1CA\s0
+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.)
+.Sp
+This is disabled by default since it requires a \s-1CA\s0 database to be
+provided by the caller in many cases.
+.IP "\fBcert_file, cert=\fR\fIfilename\fR" 4
+.IX Item "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.)
+.IP "\fBkey_file, key=\fR\fIfilename\fR" 4
+.IX Item "key_file, key=filename"
+A file containing the private key for the certificate.
+.IP "\fBlisten=\fR\fI1|0\fR" 4
+.IX Item "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.
+.PP
+Example command lines:
+.PP
+To create a \s-1TLS/SSL\s0 server that serves an input stream.
+.PP
+.Vb 1
+\& ffmpeg \-i <input> \-f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>
+.Ve
+.PP
+To play back a stream from the \s-1TLS/SSL\s0 server using \fBffplay\fR:
+.PP
+.Vb 1
+\& ffplay tls://<hostname>:<port>
+.Ve
+.SS "udp"
+.IX Subsection "udp"
+User Datagram Protocol.
+.PP
+The required syntax for an \s-1UDP\s0 \s-1URL\s0 is:
+.PP
+.Vb 1
+\& udp://<hostname>:<port>[?<options>]
+.Ve
+.PP
+\&\fIoptions\fR contains a list of &\-separated options of the form \fIkey\fR=\fIval\fR.
+.PP
+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
+\&\s-1UDP\s0 socket buffer overruns. The \fIfifo_size\fR and
+\&\fIoverrun_nonfatal\fR options are related to this buffer.
+.PP
+The list of supported options follows.
+.IP "\fBbuffer_size=\fR\fIsize\fR" 4
+.IX Item "buffer_size=size"
+Set the \s-1UDP\s0 maximum socket buffer size in bytes. This is used to set either
+the receive or send buffer size, depending on what the socket is used for.
+Default is 64KB. See also \fIfifo_size\fR.
+.IP "\fBbitrate=\fR\fIbitrate\fR" 4
+.IX Item "bitrate=bitrate"
+If set to nonzero, the output will have the specified constant bitrate if the
+input has enough packets to sustain it.
+.IP "\fBburst_bits=\fR\fIbits\fR" 4
+.IX Item "burst_bits=bits"
+When using \fIbitrate\fR this specifies the maximum number of bits in
+packet bursts.
+.IP "\fBlocalport=\fR\fIport\fR" 4
+.IX Item "localport=port"
+Override the local \s-1UDP\s0 port to bind with.
+.IP "\fBlocaladdr=\fR\fIaddr\fR" 4
+.IX Item "localaddr=addr"
+Local \s-1IP\s0 address of a network interface used for sending packets or joining
+multicast groups.
+.IP "\fBpkt_size=\fR\fIsize\fR" 4
+.IX Item "pkt_size=size"
+Set the size in bytes of \s-1UDP\s0 packets.
+.IP "\fBreuse=\fR\fI1|0\fR" 4
+.IX Item "reuse=1|0"
+Explicitly allow or disallow reusing \s-1UDP\s0 sockets.
+.IP "\fBttl=\fR\fIttl\fR" 4
+.IX Item "ttl=ttl"
+Set the time to live value (for multicast only).
+.IP "\fBconnect=\fR\fI1|0\fR" 4
+.IX Item "connect=1|0"
+Initialize the \s-1UDP\s0 socket with \f(CW\*(C`connect()\*(C'\fR. 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 \s-1AVERROR\s0(\s-1ECONNREFUSED\s0) if \*(L"destination
+unreachable\*(R" is received.
+For receiving, this gives the benefit of only receiving packets from
+the specified peer address/port.
+.IP "\fBsources=\fR\fIaddress\fR\fB[,\fR\fIaddress\fR\fB]\fR" 4
+.IX Item "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.
+.IP "\fBblock=\fR\fIaddress\fR\fB[,\fR\fIaddress\fR\fB]\fR" 4
+.IX Item "block=address[,address]"
+Ignore packets sent from the specified addresses. In case of multicast, also
+exclude the source addresses in the multicast subscription.
+.IP "\fBfifo_size=\fR\fIunits\fR" 4
+.IX Item "fifo_size=units"
+Set the \s-1UDP\s0 receiving circular buffer size, expressed as a number of
+packets with size of 188 bytes. If not specified defaults to 7*4096.
+.IP "\fBoverrun_nonfatal=\fR\fI1|0\fR" 4
+.IX Item "overrun_nonfatal=1|0"
+Survive in case of \s-1UDP\s0 receiving circular buffer overrun. Default
+value is 0.
+.IP "\fBtimeout=\fR\fImicroseconds\fR" 4
+.IX Item "timeout=microseconds"
+Set raise error timeout, expressed in microseconds.
+.Sp
+This option is only relevant in read mode: if no data arrived in more
+than this time interval, raise error.
+.IP "\fBbroadcast=\fR\fI1|0\fR" 4
+.IX Item "broadcast=1|0"
+Explicitly allow or disallow \s-1UDP\s0 broadcasting.
+.Sp
+Note that broadcasting may not work properly on networks having
+a broadcast storm protection.
+.PP
+\fIExamples\fR
+.IX Subsection "Examples"
+.IP "\(bu" 4
+Use \fBffmpeg\fR to stream over \s-1UDP\s0 to a remote endpoint:
+.Sp
+.Vb 1
+\& ffmpeg \-i <input> \-f <format> udp://<hostname>:<port>
+.Ve
+.IP "\(bu" 4
+Use \fBffmpeg\fR to stream in mpegts format over \s-1UDP\s0 using 188
+sized \s-1UDP\s0 packets, using a large input buffer:
+.Sp
+.Vb 1
+\& ffmpeg \-i <input> \-f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535
+.Ve
+.IP "\(bu" 4
+Use \fBffmpeg\fR to receive over \s-1UDP\s0 from a remote endpoint:
+.Sp
+.Vb 1
+\& ffmpeg \-i udp://[<multicast\-address>]:<port> ...
+.Ve
+.SS "unix"
+.IX Subsection "unix"
+Unix local socket
+.PP
+The required syntax for a Unix socket \s-1URL\s0 is:
+.PP
+.Vb 1
+\& unix://<filepath>
+.Ve
+.PP
+The following parameters can be set via command line options
+(or in code via \f(CW\*(C`AVOption\*(C'\fRs):
+.IP "\fBtimeout\fR" 4
+.IX Item "timeout"
+Timeout in ms.
+.IP "\fBlisten\fR" 4
+.IX Item "listen"
+Create the Unix socket in listening mode.
+.SH "SEE ALSO"
+.IX Header "SEE ALSO"
+\&\fIffmpeg\fR\|(1), \fIffplay\fR\|(1), \fIffprobe\fR\|(1), \fIlibavformat\fR\|(3)
+.SH "AUTHORS"
+.IX Header "AUTHORS"
+The FFmpeg developers.
+.PP
+For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+\&\fBgit log\fR in the FFmpeg source directory, or browsing the
+online repository at <\fBhttp://source.ffmpeg.org\fR>.
+.PP
+Maintainers for the specific components are listed in the file
+\&\fI\s-1MAINTAINERS\s0\fR in the source code tree.
--
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