Hu Chunming / FFNvDecoder

  @chapter Protocol Options
  @c man begin 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 -@var{option} @var{value} in the
  FFmpeg tools, or by setting the value explicitly in the
  @code{AVFormatContext} options or using the @file{libavutil/opt.h} API
  for programmatic use.
  
  The list of supported options follows:
  
  @table @option
  @item protocol_whitelist @var{list} (@emph{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.
  @end table
  
  @c man end PROTOCOL OPTIONS
  
  @chapter Protocols
  @c man begin 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=@var{PROTOCOL}", or you can disable a
  particular protocol using the option
  "--disable-protocol=@var{PROTOCOL}".
  
  The option "-protocols" of the ff* tools will display the list of
  supported protocols.
  
  All protocols accept the following options:
  
  @table @option
  @item rw_timeout
  Maximum time to wait for (network) read/write operations to complete,
  in microseconds.
  @end table
  
  A description of the currently available protocols follows.
  
  @section async
  
  Asynchronous data filling wrapper for input stream.
  
  Fill data in a background thread, to decouple I/O operation from demux thread.
  
  @example
  async:@var{URL}
  async:http://host/resource
  async:cache:http://host/resource
  @end example
  
  @section bluray
  
  Read BluRay playlist.
  
  The accepted options are:
  @table @option
  
  @item angle
  BluRay angle
  
  @item chapter
  Start chapter (1...N)
  
  @item playlist
  Playlist to read (BDMV/PLAYLIST/?????.mpls)
  
  @end table
  
  Examples:
  
  Read longest playlist from BluRay mounted to /mnt/bluray:
  @example
  bluray:/mnt/bluray
  @end example
  
  Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
  @example
  -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
  @end example
  
  @section cache
  
  Caching wrapper for input stream.
  
  Cache the input stream to temporary file. It brings seeking capability to live streams.
  
  @example
  cache:@var{URL}
  @end example
  
  @section 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:
  @example
  concat:@var{URL1}|@var{URL2}|...|@var{URLN}
  @end example
  
  where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
  resource to be concatenated, each one possibly specifying a distinct
  protocol.
  
  For example to read a sequence of files @file{split1.mpeg},
  @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
  command:
  @example
  ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
  @end example
  
  Note that you may need to escape the character "|" which is special for
  many shells.
  
  @section crypto
  
  AES-encrypted stream reading protocol.
  
  The accepted options are:
  @table @option
  @item key
  Set the AES decryption key binary block from given hexadecimal representation.
  
  @item iv
  Set the AES decryption initialization vector binary block from given hexadecimal representation.
  @end table
  
  Accepted URL formats:
  @example
  crypto:@var{URL}
  crypto+@var{URL}
  @end example
  
  @section data
  
  Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
  
  For example, to convert a GIF file given inline with @command{ffmpeg}:
  @example
  ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
  @end example
  
  @section file
  
  File access protocol.
  
  Read from or write to a file.
  
  A file URL can have the form:
  @example
  file:@var{filename}
  @end example
  
  where @var{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 @file{input.mpeg} with @command{ffmpeg}
  use the command:
  @example
  ffmpeg -i file:input.mpeg output.mpeg
  @end example
  
  This protocol accepts the following options:
  
  @table @option
  @item truncate
  Truncate existing files on write, if set to 1. A value of 0 prevents
  truncating. Default value is 1.
  
  @item blocksize
  Set I/O operation maximum block size, in bytes. Default value is
  @code{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.
  
  @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 API users).
  
  @item 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).
  @end table
  
  @section ftp
  
  FTP (File Transfer Protocol).
  
  Read from or write to remote resources using FTP protocol.
  
  Following syntax is required.
  @example
  ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
  @end example
  
  This protocol accepts the following options.
  
  @table @option
  @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.
  
  @item ftp-anonymous-password
  Password used when login as anonymous user. Typically an e-mail address
  should be used.
  
  @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.
  @end table
  
  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.
  
  @section gopher
  
  Gopher protocol.
  
  @section hls
  
  Read Apple HTTP Live Streaming compliant segmented stream as
  a uniform one. The M3U8 playlists describing the segments can be
  remote HTTP resources or local files, accessed using the standard
  file protocol.
  The nested protocol is declared by specifying
  "+@var{proto}" after the hls URI scheme name, where @var{proto}
  is either "file" or "http".
  
  @example
  hls+http://host/path/to/remote/resource.m3u8
  hls+file://path/to/local/resource.m3u8
  @end example
  
  Using this protocol is discouraged - the hls demuxer should work
  just as well (if not, please report the issues) and is more complete.
  To use the hls demuxer instead, simply use the direct URLs to the
  m3u8 files.
  
  @section http
  
  HTTP (Hyper Text Transfer Protocol).
  
  This protocol accepts the following options:
  
  @table @option
  @item seekable
  Control seekability of connection. If set to 1 the resource is
  supposed to be seekable, if set to 0 it is assumed not to be seekable,
  if set to -1 it will try to autodetect if it is seekable. Default
  value is -1.
  
  @item chunked_post
  If set to 1 use chunked Transfer-Encoding for posts, default is 1.
  
  @item content_type
  Set a specific content type for the POST messages or for listen mode.
  
  @item http_proxy
  set HTTP proxy to tunnel through e.g. http://example.com:1234
  
  @item headers
  Set custom HTTP headers, can override built in default headers. The
  value must be a string encoding the headers.
  
  @item multiple_requests
  Use persistent connections if set to 1, default is 0.
  
  @item post_data
  Set custom HTTP post data.
  
  @item referer
  Set the Referer header. Include 'Referer: URL' header in HTTP request.
  
  @item user_agent
  Override the User-Agent header. If not specified the protocol will use a
  string describing the libavformat build. ("Lavf/<version>")
  
  @item user-agent
  This is a deprecated option, you can use user_agent instead it.
  
  @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.
  
  @item reconnect_at_eof
  If set then eof is treated like an error and causes reconnection, this is useful
  for live / endless streams.
  
  @item reconnect_streamed
  If set then even streamed/non seekable streams will be reconnected on errors.
  
  @item reconnect_delay_max
  Sets the maximum delay in seconds after which to give up reconnecting
  
  @item mime_type
  Export the MIME type.
  
  @item http_version
  Exports the HTTP response version number. Usually "1.0" or "1.1".
  
  @item 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 @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
  The default is 1.
  
  @item icy_metadata_headers
  If the server supports ICY metadata, this contains the ICY-specific HTTP reply
  headers, separated by newline characters.
  
  @item icy_metadata_packet
  If the server supports ICY metadata, and @option{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.
  
  @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 HTTP response field. Multiple cookies can be
  delimited by a newline character.
  
  @item offset
  Set initial byte offset.
  
  @item end_offset
  Try to limit the request to bytes preceding this offset.
  
  @item 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.
  
  @item 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.
  @example
  # Server side (sending):
  ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
  
  # Client side (receiving):
  ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
  
  # Client can also be done with wget:
  wget http://@var{server}:@var{port} -O somefile.ogg
  
  # Server side (receiving):
  ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
  
  # Client side (sending):
  ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
  
  # Client can also be done with wget:
  wget --post-file=somefile.ogg http://@var{server}:@var{port}
  @end example
  
  @item 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.
  
  @end table
  
  @subsection HTTP Cookies
  
  Some HTTP requests will be denied unless cookie values are passed in with the
  request. The @option{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:
  @example
  ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
  @end example
  
  @section Icecast
  
  Icecast protocol (stream to Icecast servers)
  
  This protocol accepts the following options:
  
  @table @option
  @item ice_genre
  Set the stream genre.
  
  @item ice_name
  Set the stream name.
  
  @item ice_description
  Set the stream description.
  
  @item ice_url
  Set the stream website URL.
  
  @item ice_public
  Set if the stream should be public.
  The default is 0 (not public).
  
  @item user_agent
  Override the User-Agent header. If not specified a string of the form
  "Lavf/<version>" will be used.
  
  @item password
  Set the Icecast mountpoint password.
  
  @item content_type
  Set the stream content type. This must be set if it is different from
  audio/mpeg.
  
  @item legacy_icecast
  This enables support for Icecast versions < 2.4.0, that do not support the
  HTTP PUT method but the SOURCE method.
  
  @end table
  
  @example
  icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
  @end example
  
  @section mmst
  
  MMS (Microsoft Media Server) protocol over TCP.
  
  @section mmsh
  
  MMS (Microsoft Media Server) protocol over HTTP.
  
  The required syntax is:
  @example
  mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
  @end example
  
  @section md5
  
  MD5 output protocol.
  
  Computes the MD5 hash of the data to be written, and on close writes
  this to the designated output or stdout if none is specified. It can
  be used to test muxers without writing an actual file.
  
  Some examples follow.
  @example
  # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
  ffmpeg -i input.flv -f avi -y md5:output.avi.md5
  
  # Write the MD5 hash of the encoded AVI file to stdout.
  ffmpeg -i input.flv -f avi -y md5:
  @end example
  
  Note that some formats (typically MOV) require the output protocol to
  be seekable, so they will fail with the MD5 output protocol.
  
  @section pipe
  
  UNIX pipe access protocol.
  
  Read and write from UNIX pipes.
  
  The accepted syntax is:
  @example
  pipe:[@var{number}]
  @end example
  
  @var{number} is the number corresponding to the file descriptor of the
  pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).  If @var{number}
  is not specified, by default the stdout file descriptor will be used
  for writing, stdin for reading.
  
  For example to read from stdin with @command{ffmpeg}:
  @example
  cat test.wav | ffmpeg -i pipe:0
  # ...this is the same as...
  cat test.wav | ffmpeg -i pipe:
  @end example
  
  For writing to stdout with @command{ffmpeg}:
  @example
  ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
  # ...this is the same as...
  ffmpeg -i test.wav -f avi pipe: | cat > test.avi
  @end example
  
  This protocol accepts the following options:
  
  @table @option
  @item blocksize
  Set I/O operation maximum block size, in bytes. Default value is
  @code{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.
  @end table
  
  Note that some formats (typically MOV), require the output protocol to
  be seekable, so they will fail with the pipe output protocol.
  
  @section 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 @code{rtp_mpegts} muxer and
  the @code{rtp} protocol.
  
  The required syntax is:
  @example
  -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
  @end example
  
  The destination UDP ports are @code{port + 2} for the column FEC stream
  and @code{port + 4} for the row FEC stream.
  
  This protocol accepts the following options:
  @table @option
  
  @item l=@var{n}
  The number of columns (4-20, LxD <= 100)
  
  @item d=@var{n}
  The number of rows (4-20, LxD <= 100)
  
  @end table
  
  Example usage:
  
  @example
  -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
  @end example
  
  @section rtmp
  
  Real-Time Messaging Protocol.
  
  The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
  content across a TCP/IP network.
  
  The required syntax is:
  @example
  rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
  @end example
  
  The accepted parameters are:
  @table @option
  
  @item username
  An optional username (mostly for publishing).
  
  @item password
  An optional password (mostly for publishing).
  
  @item server
  The address of the RTMP server.
  
  @item port
  The number of the TCP port to use (by default is 1935).
  
  @item app
  It is the name of the application to access. It usually corresponds to
  the path where the application is installed on the RTMP server
  (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
  the value parsed from the URI through the @code{rtmp_app} option, too.
  
  @item playpath
  It is the path or name of the resource to play with reference to the
  application specified in @var{app}, may be prefixed by "mp4:". You
  can override the value parsed from the URI through the @code{rtmp_playpath}
  option, too.
  
  @item listen
  Act as a server, listening for an incoming connection.
  
  @item timeout
  Maximum time to wait for the incoming connection. Implies listen.
  @end table
  
  Additionally, the following parameters can be set via command line options
  (or in code via @code{AVOption}s):
  @table @option
  
  @item rtmp_app
  Name of application to connect on the RTMP server. This option
  overrides the parameter specified in the URI.
  
  @item rtmp_buffer
  Set the client buffer time in milliseconds. The default is 3000.
  
  @item rtmp_conn
  Extra arbitrary AMF connection parameters, parsed from a string,
  e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
  Each value is prefixed by a single character denoting the type,
  B for Boolean, N for number, S for string, O for object, or Z for null,
  followed by a colon. For Booleans the data must be either 0 or 1 for
  FALSE or TRUE, respectively.  Likewise for Objects the data must be 0 or
  1 to end or begin an object, respectively. Data items in subobjects may
  be named, by prefixing the type with 'N' and specifying the name before
  the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
  times to construct arbitrary AMF sequences.
  
  @item rtmp_flashver
  Version of the Flash plugin used to run the SWF player. The default
  is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
  <libavformat version>).)
  
  @item rtmp_flush_interval
  Number of packets flushed in the same request (RTMPT only). The default
  is 10.
  
  @item rtmp_live
  Specify that the media is a live stream. No resuming or seeking in
  live streams is possible. The default value is @code{any}, which means the
  subscriber first tries to play the live stream specified in the
  playpath. If a live stream of that name is not found, it plays the
  recorded stream. The other possible values are @code{live} and
  @code{recorded}.
  
  @item rtmp_pageurl
  URL of the web page in which the media was embedded. By default no
  value will be sent.
  
  @item rtmp_playpath
  Stream identifier to play or to publish. This option overrides the
  parameter specified in the URI.
  
  @item rtmp_subscribe
  Name of live stream to subscribe to. By default no value will be sent.
  It is only sent if the option is specified or if rtmp_live
  is set to live.
  
  @item rtmp_swfhash
  SHA256 hash of the decompressed SWF file (32 bytes).
  
  @item rtmp_swfsize
  Size of the decompressed SWF file, required for SWFVerification.
  
  @item rtmp_swfurl
  URL of the SWF player for the media. By default no value will be sent.
  
  @item rtmp_swfverify
  URL to player swf file, compute hash/size automatically.
  
  @item rtmp_tcurl
  URL of the target stream. Defaults to proto://host[:port]/app.
  
  @end table
  
  For example to read with @command{ffplay} a multimedia resource named
  "sample" from the application "vod" from an RTMP server "myserver":
  @example
  ffplay rtmp://myserver/vod/sample
  @end example
  
  To publish to a password protected server, passing the playpath and
  app names separately:
  @example
  ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
  @end example
  
  @section rtmpe
  
  Encrypted Real-Time Messaging Protocol.
  
  The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
  streaming multimedia content within standard cryptographic primitives,
  consisting of Diffie-Hellman key exchange and HMACSHA256, generating
  a pair of RC4 keys.
  
  @section rtmps
  
  Real-Time Messaging Protocol over a secure SSL connection.
  
  The Real-Time Messaging Protocol (RTMPS) is used for streaming
  multimedia content across an encrypted connection.
  
  @section rtmpt
  
  Real-Time Messaging Protocol tunneled through HTTP.
  
  The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
  for streaming multimedia content within HTTP requests to traverse
  firewalls.
  
  @section rtmpte
  
  Encrypted Real-Time Messaging Protocol tunneled through HTTP.
  
  The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
  is used for streaming multimedia content within HTTP requests to traverse
  firewalls.
  
  @section rtmpts
  
  Real-Time Messaging Protocol tunneled through HTTPS.
  
  The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
  for streaming multimedia content within HTTPS requests to traverse
  firewalls.
  
  @section libsmbclient
  
  libsmbclient permits one to manipulate CIFS/SMB network resources.
  
  Following syntax is required.
  
  @example
  smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
  @end example
  
  This protocol accepts the following options.
  
  @table @option
  @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.
  
  @item truncate
  Truncate existing files on write, if set to 1. A value of 0 prevents
  truncating. Default value is 1.
  
  @item workgroup
  Set the workgroup used for making connections. By default workgroup is not specified.
  
  @end table
  
  For more information see: @url{http://www.samba.org/}.
  
  @section libssh
  
  Secure File Transfer Protocol via libssh
  
  Read from or write to remote resources using SFTP protocol.
  
  Following syntax is required.
  
  @example
  sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
  @end example
  
  This protocol accepts the following options.
  
  @table @option
  @item timeout
  Set timeout of socket I/O operations used by the underlying low level
  operation. By default it is set to -1, which means that the timeout
  is not specified.
  
  @item truncate
  Truncate existing files on write, if set to 1. A value of 0 prevents
  truncating. Default value is 1.
  
  @item private_key
  Specify the path of the file containing private key to use during authorization.
  By default libssh searches for keys in the @file{~/.ssh/} directory.
  
  @end table
  
  Example: Play a file stored on remote server.
  
  @example
  ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
  @end example
  
  @section 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:
  @example
  @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
  @end example
  
  where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
  "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
  @var{server}, @var{port}, @var{app} and @var{playpath} have the same
  meaning as specified for the RTMP native protocol.
  @var{options} contains a list of space-separated options of the form
  @var{key}=@var{val}.
  
  See the librtmp manual page (man 3 librtmp) for more information.
  
  For example, to stream a file in real-time to an RTMP server using
  @command{ffmpeg}:
  @example
  ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
  @end example
  
  To play the same stream using @command{ffplay}:
  @example
  ffplay "rtmp://myserver/live/mystream live=1"
  @end example
  
  @section rtp
  
  Real-time Transport Protocol.
  
  The required syntax for an RTP URL is:
  rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
  
  @var{port} specifies the RTP port to use.
  
  The following URL options are supported:
  
  @table @option
  
  @item ttl=@var{n}
  Set the TTL (Time-To-Live) value (for multicast only).
  
  @item rtcpport=@var{n}
  Set the remote RTCP port to @var{n}.
  
  @item localrtpport=@var{n}
  Set the local RTP port to @var{n}.
  
  @item localrtcpport=@var{n}'
  Set the local RTCP port to @var{n}.
  
  @item pkt_size=@var{n}
  Set max packet size (in bytes) to @var{n}.
  
  @item connect=0|1
  Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
  to 0).
  
  @item sources=@var{ip}[,@var{ip}]
  List allowed source IP addresses.
  
  @item block=@var{ip}[,@var{ip}]
  List disallowed (blocked) source IP addresses.
  
  @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).
  
  @item localport=@var{n}
  Set the local RTP port to @var{n}.
  
  This is a deprecated option. Instead, @option{localrtpport} should be
  used.
  
  @end table
  
  Important notes:
  
  @enumerate
  
  @item
  If @option{rtcpport} is not set the RTCP port will be set to the RTP
  port value plus 1.
  
  @item
  If @option{localrtpport} (the local RTP port) is not set any available
  port will be used for the local RTP and RTCP ports.
  
  @item
  If @option{localrtcpport} (the local RTCP port) is not set it will be
  set to the local RTP port value plus 1.
  @end enumerate
  
  @section 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
  @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
  
  The required syntax for a RTSP url is:
  @example
  rtsp://@var{hostname}[:@var{port}]/@var{path}
  @end example
  
  Options can be set on the @command{ffmpeg}/@command{ffplay} command
  line, or set in code via @code{AVOption}s or in
  @code{avformat_open_input}.
  
  The following options are supported.
  
  @table @option
  @item initial_pause
  Do not start playing the stream immediately if set to 1. Default value
  is 0.
  
  @item rtsp_transport
  Set RTSP transport protocols.
  
  It accepts the following values:
  @table @samp
  @item udp
  Use UDP as lower transport protocol.
  
  @item tcp
  Use TCP (interleaving within the RTSP control channel) as lower
  transport protocol.
  
  @item udp_multicast
  Use UDP multicast as lower transport protocol.
  
  @item http
  Use HTTP tunneling as lower transport protocol, which is useful for
  passing proxies.
  @end table
  
  Multiple lower transport protocols may be specified, in that case they are
  tried one at a time (if the setup of one fails, the next one is tried).
  For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
  
  @item rtsp_flags
  Set RTSP flags.
  
  The following values are accepted:
  @table @samp
  @item filter_src
  Accept packets only from negotiated peer address and port.
  @item listen
  Act as a server, listening for an incoming connection.
  @item prefer_tcp
  Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
  @end table
  
  Default value is @samp{none}.
  
  @item allowed_media_types
  Set media types to accept from the server.
  
  The following flags are accepted:
  @table @samp
  @item video
  @item audio
  @item data
  @end table
  
  By default it accepts all media types.
  
  @item min_port
  Set minimum local UDP port. Default value is 5000.
  
  @item max_port
  Set maximum local UDP port. Default value is 65000.
  
  @item timeout
  Set maximum timeout (in seconds) to wait for incoming connections.
  
  A value of -1 means infinite (default). This option implies the
  @option{rtsp_flags} set to @samp{listen}.
  
  @item reorder_queue_size
  Set number of packets to buffer for handling of reordered packets.
  
  @item stimeout
  Set socket TCP I/O timeout in microseconds.
  
  @item user-agent
  Override User-Agent header. If not specified, it defaults to the
  libavformat identifier string.
  @end table
  
  When receiving data over UDP, the demuxer tries to reorder received packets
  (since they may arrive out of order, or packets may get lost totally). This
  can be disabled by setting the maximum demuxing delay to zero (via
  the @code{max_delay} field of AVFormatContext).
  
  When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
  streams to display can be chosen with @code{-vst} @var{n} and
  @code{-ast} @var{n} for video and audio respectively, and can be switched
  on the fly by pressing @code{v} and @code{a}.
  
  @subsection Examples
  
  The following examples all make use of the @command{ffplay} and
  @command{ffmpeg} tools.
  
  @itemize
  @item
  Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
  @example
  ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
  @end example
  
  @item
  Watch a stream tunneled over HTTP:
  @example
  ffplay -rtsp_transport http rtsp://server/video.mp4
  @end example
  
  @item
  Send a stream in realtime to a RTSP server, for others to watch:
  @example
  ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
  @end example
  
  @item
  Receive a stream in realtime:
  @example
  ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
  @end example
  @end itemize
  
  @section sap
  
  Session Announcement Protocol (RFC 2974). This is not technically a
  protocol handler in libavformat, it is a muxer and demuxer.
  It is used for signalling of RTP streams, by announcing the SDP for the
  streams regularly on a separate port.
  
  @subsection Muxer
  
  The syntax for a SAP url given to the muxer is:
  @example
  sap://@var{destination}[:@var{port}][?@var{options}]
  @end example
  
  The RTP packets are sent to @var{destination} on port @var{port},
  or to port 5004 if no port is specified.
  @var{options} is a @code{&}-separated list. The following options
  are supported:
  
  @table @option
  
  @item announce_addr=@var{address}
  Specify the destination IP address for sending the announcements to.
  If omitted, the announcements are sent to the commonly used SAP
  announcement multicast address 224.2.127.254 (sap.mcast.net), or
  ff0e::2:7ffe if @var{destination} is an IPv6 address.
  
  @item announce_port=@var{port}
  Specify the port to send the announcements on, defaults to
  9875 if not specified.
  
  @item ttl=@var{ttl}
  Specify the time to live value for the announcements and RTP packets,
  defaults to 255.
  
  @item same_port=@var{0|1}
  If set to 1, send all RTP streams on the same port pair. If zero (the
  default), all streams are sent on unique ports, with each stream on a
  port 2 numbers higher than the previous.
  VLC/Live555 requires this to be set to 1, to be able to receive the stream.
  The RTP stack in libavformat for receiving requires all streams to be sent
  on unique ports.
  @end table
  
  Example command lines follow.
  
  To broadcast a stream on the local subnet, for watching in VLC:
  
  @example
  ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
  @end example
  
  Similarly, for watching in @command{ffplay}:
  
  @example
  ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
  @end example
  
  And for watching in @command{ffplay}, over IPv6:
  
  @example
  ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
  @end example
  
  @subsection Demuxer
  
  The syntax for a SAP url given to the demuxer is:
  @example
  sap://[@var{address}][:@var{port}]
  @end example
  
  @var{address} is the multicast address to listen for announcements on,
  if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
  is the port that is listened on, 9875 if omitted.
  
  The demuxers listens for announcements on the given address and port.
  Once an announcement is received, it tries to receive that particular stream.
  
  Example command lines follow.
  
  To play back the first stream announced on the normal SAP multicast address:
  
  @example
  ffplay sap://
  @end example
  
  To play back the first stream announced on one the default IPv6 SAP multicast address:
  
  @example
  ffplay sap://[ff0e::2:7ffe]
  @end example
  
  @section sctp
  
  Stream Control Transmission Protocol.
  
  The accepted URL syntax is:
  @example
  sctp://@var{host}:@var{port}[?@var{options}]
  @end example
  
  The protocol accepts the following options:
  @table @option
  @item listen
  If set to any value, listen for an incoming connection. Outgoing connection is done by default.
  
  @item max_streams
  Set the maximum number of streams. By default no limit is set.
  @end table
  
  @section srt
  
  Haivision Secure Reliable Transport Protocol via libsrt.
  
  The supported syntax for a SRT URL is:
  @example
  srt://@var{hostname}:@var{port}[?@var{options}]
  @end example
  
  @var{options} contains a list of &-separated options of the form
  @var{key}=@var{val}.
  
  or
  
  @example
  @var{options} srt://@var{hostname}:@var{port}
  @end example
  
  @var{options} contains a list of '-@var{key} @var{val}'
  options.
  
  This protocol accepts the following options.
  
  @table @option
  @item connect_timeout
  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).
  
  @item ffs=@var{bytes}
  Flight Flag Size (Window Size), in bytes. FFS is actually an
  internal parameter and you should set it to not less than
  @option{recv_buffer_size} and @option{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.
  
  @item inputbw=@var{bytes/seconds}
  Sender nominal input rate, in bytes per seconds. Used along with
  @option{oheadbw}, when @option{maxbw} is set to relative (0), to
  calculate maximum sending rate when recovery packets are sent
  along with the main media stream:
  @option{inputbw} * (100 + @option{oheadbw}) / 100
  if @option{inputbw} is not set while @option{maxbw} is set to
  relative (0), the actual input rate is evaluated inside
  the library. Default value is 0.
  
  @item iptos=@var{tos}
  IP Type of Service. Applies to sender only. Default value is 0xB8.
  
  @item ipttl=@var{ttl}
  IP Time To Live. Applies to sender only. Default value is 64.
  
  @item latency
  Timestamp-based Packet Delivery Delay.
  Used to absorb bursts of missed packet retransmissions.
  This flag sets both @option{rcvlatency} and @option{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 @option{peerlatency},
  when side is sender and @option{rcvlatency}
  when side is receiver, and the bidirectional stream
  sending is not supported.
  
  @item listen_timeout
  Set socket listen timeout.
  
  @item maxbw=@var{bytes/seconds}
  Maximum sending bandwidth, in bytes per seconds.
  -1 infinite (CSRTCC limit is 30mbps)
  0 relative to input rate (see @option{inputbw})
  >0 absolute limit value
  Default value is 0 (relative)
  
  @item mode=@var{caller|listener|rendezvous}
  Connection mode.
  @option{caller} opens client connection.
  @option{listener} starts server to listen for incoming connections.
  @option{rendezvous} use Rendez-Vous connection mode.
  Default value is caller.
  
  @item mss=@var{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.
  
  @item nakreport=@var{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.
  
  @item oheadbw=@var{percents}
  Recovery bandwidth overhead above input rate, in percents.
  See @option{inputbw}. Default value is 25%.
  
  @item passphrase=@var{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 @option{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).
  
  @item payload_size=@var{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.
  
  @item pkt_size=@var{bytes}
  Alias for @samp{payload_size}.
  
  @item peerlatency
  The latency value (as described in @option{rcvlatency}) that is
  set by the sender side as a minimum value for the receiver.
  
  @item pbkeylen=@var{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.
  
  @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 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 @option{peerlatency}
  set by the peer side. Before version 1.3.0 this option
  is only available as @option{latency}.
  
  @item recv_buffer_size=@var{bytes}
  Set UDP receive buffer size, expressed in bytes.
  
  @item send_buffer_size=@var{bytes}
  Set UDP send buffer size, expressed in bytes.
  
  @item rw_timeout
  Set raise error timeout for read/write optations.
  
  This option is only relevant in read mode:
  if no data arrived in more than this time
  interval, raise error.
  
  @item tlpktdrop=@var{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.
  
  @item sndbuf=@var{bytes}
  Set send buffer size, expressed in bytes.
  
  @item rcvbuf=@var{bytes}
  Set receive buffer size, expressed in bytes.
  
  Receive buffer must not be greater than @option{ffs}.
  
  @item lossmaxttl=@var{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.
  
  @item 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.
  
  @item streamid=@var{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 doesn’t make sense in Rendezvous connection; the result
  might be that simply one side will override the value from the other
  side and it’s the matter of luck which one would win
  
  @item smoother=@var{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.
  
  @item messageapi=@var{1|0}
  When set, this socket uses the Message API, otherwise it uses Buffer
  API. Note that in live mode (see @option{transtype}) there’s 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.
  
  @item transtype=@var{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 @option{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 @option{messageapi}
  for further explanations
  
  @end table
  
  For more information see: @url{https://github.com/Haivision/srt}.
  
  @section srtp
  
  Secure Real-time Transport Protocol.
  
  The accepted options are:
  @table @option
  @item srtp_in_suite
  @item srtp_out_suite
  Select input and output encoding suites.
  
  Supported values:
  @table @samp
  @item AES_CM_128_HMAC_SHA1_80
  @item SRTP_AES128_CM_HMAC_SHA1_80
  @item AES_CM_128_HMAC_SHA1_32
  @item SRTP_AES128_CM_HMAC_SHA1_32
  @end table
  
  @item srtp_in_params
  @item 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.
  @end table
  
  @section subfile
  
  Virtually extract a segment of a file or another stream.
  The underlying stream must be seekable.
  
  Accepted options:
  @table @option
  @item start
  Start offset of the extracted segment, in bytes.
  @item end
  End offset of the extracted segment, in bytes.
  If set to 0, extract till end of file.
  @end table
  
  Examples:
  
  Extract a chapter from a DVD VOB file (start and end sectors obtained
  externally and multiplied by 2048):
  @example
  subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
  @end example
  
  Play an AVI file directly from a TAR archive:
  @example
  subfile,,start,183241728,end,366490624,,:archive.tar
  @end example
  
  Play a MPEG-TS file from start offset till end:
  @example
  subfile,,start,32815239,end,0,,:video.ts
  @end example
  
  @section tee
  
  Writes the output to multiple protocols. The individual outputs are separated
  by |
  
  @example
  tee:file://path/to/local/this.avi|file://path/to/local/that.avi
  @end example
  
  @section tcp
  
  Transmission Control Protocol.
  
  The required syntax for a TCP url is:
  @example
  tcp://@var{hostname}:@var{port}[?@var{options}]
  @end example
  
  @var{options} contains a list of &-separated options of the form
  @var{key}=@var{val}.
  
  The list of supported options follows.
  
  @table @option
  @item listen=@var{1|0}
  Listen for an incoming connection. Default value is 0.
  
  @item timeout=@var{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.
  
  @item listen_timeout=@var{milliseconds}
  Set listen timeout, expressed in milliseconds.
  
  @item recv_buffer_size=@var{bytes}
  Set receive buffer size, expressed bytes.
  
  @item send_buffer_size=@var{bytes}
  Set send buffer size, expressed bytes.
  
  @item tcp_nodelay=@var{1|0}
  Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
  
  @item tcp_mss=@var{bytes}
  Set maximum segment size for outgoing TCP packets, expressed in bytes.
  @end table
  
  The following example shows how to setup a listening TCP connection
  with @command{ffmpeg}, which is then accessed with @command{ffplay}:
  @example
  ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
  ffplay tcp://@var{hostname}:@var{port}
  @end example
  
  @section tls
  
  Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
  
  The required syntax for a TLS/SSL url is:
  @example
  tls://@var{hostname}:@var{port}[?@var{options}]
  @end example
  
  The following parameters can be set via command line options
  (or in code via @code{AVOption}s):
  
  @table @option
  
  @item ca_file, cafile=@var{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.
  
  @item tls_verify=@var{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.
  
  @item cert_file, cert=@var{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.)
  
  @item key_file, key=@var{filename}
  A file containing the private key for the certificate.
  
  @item listen=@var{1|0}
  If enabled, listen for connections on the provided port, and assume
  the server role in the handshake instead of the client role.
  
  @end table
  
  Example command lines:
  
  To create a TLS/SSL server that serves an input stream.
  
  @example
  ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
  @end example
  
  To play back a stream from the TLS/SSL server using @command{ffplay}:
  
  @example
  ffplay tls://@var{hostname}:@var{port}
  @end example
  
  @section udp
  
  User Datagram Protocol.
  
  The required syntax for an UDP URL is:
  @example
  udp://@var{hostname}:@var{port}[?@var{options}]
  @end example
  
  @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
  
  In case threading is enabled on the system, a circular buffer is used
  to store the incoming data, which allows one to reduce loss of data due to
  UDP socket buffer overruns. The @var{fifo_size} and
  @var{overrun_nonfatal} options are related to this buffer.
  
  The list of supported options follows.
  
  @table @option
  @item buffer_size=@var{size}
  Set the UDP 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 @var{fifo_size}.
  
  @item bitrate=@var{bitrate}
  If set to nonzero, the output will have the specified constant bitrate if the
  input has enough packets to sustain it.
  
  @item burst_bits=@var{bits}
  When using @var{bitrate} this specifies the maximum number of bits in
  packet bursts.
  
  @item localport=@var{port}
  Override the local UDP port to bind with.
  
  @item localaddr=@var{addr}
  Local IP address of a network interface used for sending packets or joining
  multicast groups.
  
  @item pkt_size=@var{size}
  Set the size in bytes of UDP packets.
  
  @item reuse=@var{1|0}
  Explicitly allow or disallow reusing UDP sockets.
  
  @item ttl=@var{ttl}
  Set the time to live value (for multicast only).
  
  @item connect=@var{1|0}
  Initialize the UDP socket with @code{connect()}. In this case, the
  destination address can't be changed with ff_udp_set_remote_url later.
  If the destination address isn't known at the start, this option can
  be specified in ff_udp_set_remote_url, too.
  This allows finding out the source address for the packets with getsockname,
  and makes writes return with AVERROR(ECONNREFUSED) if "destination
  unreachable" is received.
  For receiving, this gives the benefit of only receiving packets from
  the specified peer address/port.
  
  @item sources=@var{address}[,@var{address}]
  Only receive packets sent from the specified addresses. In case of multicast,
  also subscribe to multicast traffic coming from these addresses only.
  
  @item block=@var{address}[,@var{address}]
  Ignore packets sent from the specified addresses. In case of multicast, also
  exclude the source addresses in the multicast subscription.
  
  @item fifo_size=@var{units}
  Set the UDP receiving circular buffer size, expressed as a number of
  packets with size of 188 bytes. If not specified defaults to 7*4096.
  
  @item overrun_nonfatal=@var{1|0}
  Survive in case of UDP receiving circular buffer overrun. Default
  value is 0.
  
  @item timeout=@var{microseconds}
  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.
  
  @item broadcast=@var{1|0}
  Explicitly allow or disallow UDP broadcasting.
  
  Note that broadcasting may not work properly on networks having
  a broadcast storm protection.
  @end table
  
  @subsection Examples
  
  @itemize
  @item
  Use @command{ffmpeg} to stream over UDP to a remote endpoint:
  @example
  ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
  @end example
  
  @item
  Use @command{ffmpeg} to stream in mpegts format over UDP using 188
  sized UDP packets, using a large input buffer:
  @example
  ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
  @end example
  
  @item
  Use @command{ffmpeg} to receive over UDP from a remote endpoint:
  @example
  ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
  @end example
  @end itemize
  
  @section unix
  
  Unix local socket
  
  The required syntax for a Unix socket URL is:
  
  @example
  unix://@var{filepath}
  @end example
  
  The following parameters can be set via command line options
  (or in code via @code{AVOption}s):
  
  @table @option
  @item timeout
  Timeout in ms.
  @item listen
  Create the Unix socket in listening mode.
  @end table
  
  @c man end PROTOCOLS