1 @chapter Protocol Options
2 @c man begin PROTOCOL OPTIONS
4 The libavformat library provides some generic global options, which
5 can be set on all the protocols. In addition each protocol may support
6 so-called private options, which are specific for that component.
8 Options may be set by specifying -@var{option} @var{value} in the
9 FFmpeg tools, or by setting the value explicitly in the
10 @code{AVFormatContext} options or using the @file{libavutil/opt.h} API
13 The list of supported options follows:
16 @item protocol_whitelist @var{list} (@emph{input})
17 Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
18 prefixed by "-" are disabled.
19 All protocols are allowed by default but protocols used by an another
20 protocol (nested protocols) are restricted to a per protocol subset.
23 @c man end PROTOCOL OPTIONS
26 @c man begin PROTOCOLS
28 Protocols are configured elements in FFmpeg that enable access to
29 resources that require specific protocols.
31 When you configure your FFmpeg build, all the supported protocols are
32 enabled by default. You can list all available ones using the
33 configure option "--list-protocols".
35 You can disable all the protocols using the configure option
36 "--disable-protocols", and selectively enable a protocol using the
37 option "--enable-protocol=@var{PROTOCOL}", or you can disable a
38 particular protocol using the option
39 "--disable-protocol=@var{PROTOCOL}".
41 The option "-protocols" of the ff* tools will display the list of
44 All protocols accept the following options:
48 Maximum time to wait for (network) read/write operations to complete,
52 A description of the currently available protocols follows.
56 Asynchronous data filling wrapper for input stream.
58 Fill data in a background thread, to decouple I/O operation from demux thread.
62 async:http://host/resource
63 async:cache:http://host/resource
70 The accepted options are:
80 Playlist to read (BDMV/PLAYLIST/?????.mpls)
86 Read longest playlist from BluRay mounted to /mnt/bluray:
91 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
93 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
98 Caching wrapper for input stream.
100 Cache the input stream to temporary file. It brings seeking capability to live streams.
108 Physical concatenation protocol.
110 Read and seek from many resources in sequence as if they were
113 A URL accepted by this protocol has the syntax:
115 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
118 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
119 resource to be concatenated, each one possibly specifying a distinct
122 For example to read a sequence of files @file{split1.mpeg},
123 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
126 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
129 Note that you may need to escape the character "|" which is special for
134 AES-encrypted stream reading protocol.
136 The accepted options are:
139 Set the AES decryption key binary block from given hexadecimal representation.
142 Set the AES decryption initialization vector binary block from given hexadecimal representation.
145 Accepted URL formats:
153 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
155 For example, to convert a GIF file given inline with @command{ffmpeg}:
157 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
162 File access protocol.
164 Read from or write to a file.
166 A file URL can have the form:
171 where @var{filename} is the path of the file to read.
173 An URL that does not have a protocol prefix will be assumed to be a
174 file URL. Depending on the build, an URL that looks like a Windows
175 path with the drive letter at the beginning will also be assumed to be
176 a file URL (usually not the case in builds for unix-like systems).
178 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
181 ffmpeg -i file:input.mpeg output.mpeg
184 This protocol accepts the following options:
188 Truncate existing files on write, if set to 1. A value of 0 prevents
189 truncating. Default value is 1.
192 Set I/O operation maximum block size, in bytes. Default value is
193 @code{INT_MAX}, which results in not limiting the requested block size.
194 Setting this value reasonably low improves user termination request reaction
195 time, which is valuable for files on slow medium.
198 If set to 1, the protocol will retry reading at the end of the file, allowing
199 reading files that still are being written. In order for this to terminate,
200 you either need to use the rw_timeout option, or use the interrupt callback
204 Controls if seekability is advertised on the file. 0 means non-seekable, -1
205 means auto (seekable for normal files, non-seekable for named pipes).
207 Many demuxers handle seekable and non-seekable resources differently,
208 overriding this might speed up opening certain files at the cost of losing some
209 features (e.g. accurate seeking).
214 FTP (File Transfer Protocol).
216 Read from or write to remote resources using FTP protocol.
218 Following syntax is required.
220 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
223 This protocol accepts the following options.
227 Set timeout in microseconds of socket I/O operations used by the underlying low level
228 operation. By default it is set to -1, which means that the timeout is
232 Set a user to be used for authenticating to the FTP server. This is overridden by the
236 Set a password to be used for authenticating to the FTP server. This is overridden by
237 the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
239 @item ftp-anonymous-password
240 Password used when login as anonymous user. Typically an e-mail address
243 @item ftp-write-seekable
244 Control seekability of connection during encoding. If set to 1 the
245 resource is supposed to be seekable, if set to 0 it is assumed not
246 to be seekable. Default value is 0.
249 NOTE: Protocol can be used as output, but it is recommended to not do
250 it, unless special care is taken (tests, customized server configuration
251 etc.). Different FTP servers behave in different way during seek
252 operation. ff* tools may produce incomplete content due to server limitations.
260 Read Apple HTTP Live Streaming compliant segmented stream as
261 a uniform one. The M3U8 playlists describing the segments can be
262 remote HTTP resources or local files, accessed using the standard
264 The nested protocol is declared by specifying
265 "+@var{proto}" after the hls URI scheme name, where @var{proto}
266 is either "file" or "http".
269 hls+http://host/path/to/remote/resource.m3u8
270 hls+file://path/to/local/resource.m3u8
273 Using this protocol is discouraged - the hls demuxer should work
274 just as well (if not, please report the issues) and is more complete.
275 To use the hls demuxer instead, simply use the direct URLs to the
280 HTTP (Hyper Text Transfer Protocol).
282 This protocol accepts the following options:
286 Control seekability of connection. If set to 1 the resource is
287 supposed to be seekable, if set to 0 it is assumed not to be seekable,
288 if set to -1 it will try to autodetect if it is seekable. Default
292 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
295 Set a specific content type for the POST messages or for listen mode.
298 set HTTP proxy to tunnel through e.g. http://example.com:1234
301 Set custom HTTP headers, can override built in default headers. The
302 value must be a string encoding the headers.
304 @item multiple_requests
305 Use persistent connections if set to 1, default is 0.
308 Set custom HTTP post data.
311 Set the Referer header. Include 'Referer: URL' header in HTTP request.
314 Override the User-Agent header. If not specified the protocol will use a
315 string describing the libavformat build. ("Lavf/<version>")
318 This is a deprecated option, you can use user_agent instead it.
321 Set timeout in microseconds of socket I/O operations used by the underlying low level
322 operation. By default it is set to -1, which means that the timeout is
325 @item reconnect_at_eof
326 If set then eof is treated like an error and causes reconnection, this is useful
327 for live / endless streams.
329 @item reconnect_streamed
330 If set then even streamed/non seekable streams will be reconnected on errors.
332 @item reconnect_delay_max
333 Sets the maximum delay in seconds after which to give up reconnecting
336 Export the MIME type.
339 Exports the HTTP response version number. Usually "1.0" or "1.1".
342 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
343 supports this, the metadata has to be retrieved by the application by reading
344 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
347 @item icy_metadata_headers
348 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
349 headers, separated by newline characters.
351 @item icy_metadata_packet
352 If the server supports ICY metadata, and @option{icy} was set to 1, this
353 contains the last non-empty metadata packet sent by the server. It should be
354 polled in regular intervals by applications interested in mid-stream metadata
358 Set the cookies to be sent in future requests. The format of each cookie is the
359 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
360 delimited by a newline character.
363 Set initial byte offset.
366 Try to limit the request to bytes preceding this offset.
369 When used as a client option it sets the HTTP method for the request.
371 When used as a server option it sets the HTTP method that is going to be
372 expected from the client(s).
373 If the expected and the received HTTP method do not match the client will
374 be given a Bad Request response.
375 When unset the HTTP method is not checked for now. This will be replaced by
376 autodetection in the future.
379 If set to 1 enables experimental HTTP server. This can be used to send data when
380 used as an output option, or read data from a client with HTTP POST when used as
382 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
383 in ffmpeg.c and thus must not be used as a command line option.
385 # Server side (sending):
386 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
388 # Client side (receiving):
389 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
391 # Client can also be done with wget:
392 wget http://@var{server}:@var{port} -O somefile.ogg
394 # Server side (receiving):
395 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
397 # Client side (sending):
398 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
400 # Client can also be done with wget:
401 wget --post-file=somefile.ogg http://@var{server}:@var{port}
404 @item send_expect_100
405 Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
406 to 0 it won't, if set to -1 it will try to send if it is applicable. Default
411 @subsection HTTP Cookies
413 Some HTTP requests will be denied unless cookie values are passed in with the
414 request. The @option{cookies} option allows these cookies to be specified. At
415 the very least, each cookie must specify a value along with a path and domain.
416 HTTP requests that match both the domain and path will automatically include the
417 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
420 The required syntax to play a stream specifying a cookie is:
422 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
427 Icecast protocol (stream to Icecast servers)
429 This protocol accepts the following options:
433 Set the stream genre.
438 @item ice_description
439 Set the stream description.
442 Set the stream website URL.
445 Set if the stream should be public.
446 The default is 0 (not public).
449 Override the User-Agent header. If not specified a string of the form
450 "Lavf/<version>" will be used.
453 Set the Icecast mountpoint password.
456 Set the stream content type. This must be set if it is different from
460 This enables support for Icecast versions < 2.4.0, that do not support the
461 HTTP PUT method but the SOURCE method.
466 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
471 MMS (Microsoft Media Server) protocol over TCP.
475 MMS (Microsoft Media Server) protocol over HTTP.
477 The required syntax is:
479 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
486 Computes the MD5 hash of the data to be written, and on close writes
487 this to the designated output or stdout if none is specified. It can
488 be used to test muxers without writing an actual file.
490 Some examples follow.
492 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
493 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
495 # Write the MD5 hash of the encoded AVI file to stdout.
496 ffmpeg -i input.flv -f avi -y md5:
499 Note that some formats (typically MOV) require the output protocol to
500 be seekable, so they will fail with the MD5 output protocol.
504 UNIX pipe access protocol.
506 Read and write from UNIX pipes.
508 The accepted syntax is:
513 @var{number} is the number corresponding to the file descriptor of the
514 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
515 is not specified, by default the stdout file descriptor will be used
516 for writing, stdin for reading.
518 For example to read from stdin with @command{ffmpeg}:
520 cat test.wav | ffmpeg -i pipe:0
521 # ...this is the same as...
522 cat test.wav | ffmpeg -i pipe:
525 For writing to stdout with @command{ffmpeg}:
527 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
528 # ...this is the same as...
529 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
532 This protocol accepts the following options:
536 Set I/O operation maximum block size, in bytes. Default value is
537 @code{INT_MAX}, which results in not limiting the requested block size.
538 Setting this value reasonably low improves user termination request reaction
539 time, which is valuable if data transmission is slow.
542 Note that some formats (typically MOV), require the output protocol to
543 be seekable, so they will fail with the pipe output protocol.
547 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
549 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
550 for MPEG-2 Transport Streams sent over RTP.
552 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
553 the @code{rtp} protocol.
555 The required syntax is:
557 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
560 The destination UDP ports are @code{port + 2} for the column FEC stream
561 and @code{port + 4} for the row FEC stream.
563 This protocol accepts the following options:
567 The number of columns (4-20, LxD <= 100)
570 The number of rows (4-20, LxD <= 100)
577 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
582 Real-Time Messaging Protocol.
584 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
585 content across a TCP/IP network.
587 The required syntax is:
589 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
592 The accepted parameters are:
596 An optional username (mostly for publishing).
599 An optional password (mostly for publishing).
602 The address of the RTMP server.
605 The number of the TCP port to use (by default is 1935).
608 It is the name of the application to access. It usually corresponds to
609 the path where the application is installed on the RTMP server
610 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
611 the value parsed from the URI through the @code{rtmp_app} option, too.
614 It is the path or name of the resource to play with reference to the
615 application specified in @var{app}, may be prefixed by "mp4:". You
616 can override the value parsed from the URI through the @code{rtmp_playpath}
620 Act as a server, listening for an incoming connection.
623 Maximum time to wait for the incoming connection. Implies listen.
626 Additionally, the following parameters can be set via command line options
627 (or in code via @code{AVOption}s):
631 Name of application to connect on the RTMP server. This option
632 overrides the parameter specified in the URI.
635 Set the client buffer time in milliseconds. The default is 3000.
638 Extra arbitrary AMF connection parameters, parsed from a string,
639 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
640 Each value is prefixed by a single character denoting the type,
641 B for Boolean, N for number, S for string, O for object, or Z for null,
642 followed by a colon. For Booleans the data must be either 0 or 1 for
643 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
644 1 to end or begin an object, respectively. Data items in subobjects may
645 be named, by prefixing the type with 'N' and specifying the name before
646 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
647 times to construct arbitrary AMF sequences.
650 Version of the Flash plugin used to run the SWF player. The default
651 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
652 <libavformat version>).)
654 @item rtmp_flush_interval
655 Number of packets flushed in the same request (RTMPT only). The default
659 Specify that the media is a live stream. No resuming or seeking in
660 live streams is possible. The default value is @code{any}, which means the
661 subscriber first tries to play the live stream specified in the
662 playpath. If a live stream of that name is not found, it plays the
663 recorded stream. The other possible values are @code{live} and
667 URL of the web page in which the media was embedded. By default no
671 Stream identifier to play or to publish. This option overrides the
672 parameter specified in the URI.
675 Name of live stream to subscribe to. By default no value will be sent.
676 It is only sent if the option is specified or if rtmp_live
680 SHA256 hash of the decompressed SWF file (32 bytes).
683 Size of the decompressed SWF file, required for SWFVerification.
686 URL of the SWF player for the media. By default no value will be sent.
689 URL to player swf file, compute hash/size automatically.
692 URL of the target stream. Defaults to proto://host[:port]/app.
696 For example to read with @command{ffplay} a multimedia resource named
697 "sample" from the application "vod" from an RTMP server "myserver":
699 ffplay rtmp://myserver/vod/sample
702 To publish to a password protected server, passing the playpath and
703 app names separately:
705 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
710 Encrypted Real-Time Messaging Protocol.
712 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
713 streaming multimedia content within standard cryptographic primitives,
714 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
719 Real-Time Messaging Protocol over a secure SSL connection.
721 The Real-Time Messaging Protocol (RTMPS) is used for streaming
722 multimedia content across an encrypted connection.
726 Real-Time Messaging Protocol tunneled through HTTP.
728 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
729 for streaming multimedia content within HTTP requests to traverse
734 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
736 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
737 is used for streaming multimedia content within HTTP requests to traverse
742 Real-Time Messaging Protocol tunneled through HTTPS.
744 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
745 for streaming multimedia content within HTTPS requests to traverse
748 @section libsmbclient
750 libsmbclient permits one to manipulate CIFS/SMB network resources.
752 Following syntax is required.
755 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
758 This protocol accepts the following options.
762 Set timeout in milliseconds of socket I/O operations used by the underlying
763 low level operation. By default it is set to -1, which means that the timeout
767 Truncate existing files on write, if set to 1. A value of 0 prevents
768 truncating. Default value is 1.
771 Set the workgroup used for making connections. By default workgroup is not specified.
775 For more information see: @url{http://www.samba.org/}.
779 Secure File Transfer Protocol via libssh
781 Read from or write to remote resources using SFTP protocol.
783 Following syntax is required.
786 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
789 This protocol accepts the following options.
793 Set timeout of socket I/O operations used by the underlying low level
794 operation. By default it is set to -1, which means that the timeout
798 Truncate existing files on write, if set to 1. A value of 0 prevents
799 truncating. Default value is 1.
802 Specify the path of the file containing private key to use during authorization.
803 By default libssh searches for keys in the @file{~/.ssh/} directory.
807 Example: Play a file stored on remote server.
810 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
813 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
815 Real-Time Messaging Protocol and its variants supported through
818 Requires the presence of the librtmp headers and library during
819 configuration. You need to explicitly configure the build with
820 "--enable-librtmp". If enabled this will replace the native RTMP
823 This protocol provides most client functions and a few server
824 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
825 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
826 variants of these encrypted types (RTMPTE, RTMPTS).
828 The required syntax is:
830 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
833 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
834 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
835 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
836 meaning as specified for the RTMP native protocol.
837 @var{options} contains a list of space-separated options of the form
840 See the librtmp manual page (man 3 librtmp) for more information.
842 For example, to stream a file in real-time to an RTMP server using
845 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
848 To play the same stream using @command{ffplay}:
850 ffplay "rtmp://myserver/live/mystream live=1"
855 Real-time Transport Protocol.
857 The required syntax for an RTP URL is:
858 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
860 @var{port} specifies the RTP port to use.
862 The following URL options are supported:
867 Set the TTL (Time-To-Live) value (for multicast only).
869 @item rtcpport=@var{n}
870 Set the remote RTCP port to @var{n}.
872 @item localrtpport=@var{n}
873 Set the local RTP port to @var{n}.
875 @item localrtcpport=@var{n}'
876 Set the local RTCP port to @var{n}.
878 @item pkt_size=@var{n}
879 Set max packet size (in bytes) to @var{n}.
882 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
885 @item sources=@var{ip}[,@var{ip}]
886 List allowed source IP addresses.
888 @item block=@var{ip}[,@var{ip}]
889 List disallowed (blocked) source IP addresses.
891 @item write_to_source=0|1
892 Send packets to the source address of the latest received packet (if
893 set to 1) or to a default remote address (if set to 0).
895 @item localport=@var{n}
896 Set the local RTP port to @var{n}.
898 This is a deprecated option. Instead, @option{localrtpport} should be
908 If @option{rtcpport} is not set the RTCP port will be set to the RTP
912 If @option{localrtpport} (the local RTP port) is not set any available
913 port will be used for the local RTP and RTCP ports.
916 If @option{localrtcpport} (the local RTCP port) is not set it will be
917 set to the local RTP port value plus 1.
922 Real-Time Streaming Protocol.
924 RTSP is not technically a protocol handler in libavformat, it is a demuxer
925 and muxer. The demuxer supports both normal RTSP (with data transferred
926 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
927 data transferred over RDT).
929 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
930 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
931 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
933 The required syntax for a RTSP url is:
935 rtsp://@var{hostname}[:@var{port}]/@var{path}
938 Options can be set on the @command{ffmpeg}/@command{ffplay} command
939 line, or set in code via @code{AVOption}s or in
940 @code{avformat_open_input}.
942 The following options are supported.
946 Do not start playing the stream immediately if set to 1. Default value
950 Set RTSP transport protocols.
952 It accepts the following values:
955 Use UDP as lower transport protocol.
958 Use TCP (interleaving within the RTSP control channel) as lower
962 Use UDP multicast as lower transport protocol.
965 Use HTTP tunneling as lower transport protocol, which is useful for
969 Multiple lower transport protocols may be specified, in that case they are
970 tried one at a time (if the setup of one fails, the next one is tried).
971 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
976 The following values are accepted:
979 Accept packets only from negotiated peer address and port.
981 Act as a server, listening for an incoming connection.
983 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
986 Default value is @samp{none}.
988 @item allowed_media_types
989 Set media types to accept from the server.
991 The following flags are accepted:
998 By default it accepts all media types.
1001 Set minimum local UDP port. Default value is 5000.
1004 Set maximum local UDP port. Default value is 65000.
1007 Set maximum timeout (in seconds) to wait for incoming connections.
1009 A value of -1 means infinite (default). This option implies the
1010 @option{rtsp_flags} set to @samp{listen}.
1012 @item reorder_queue_size
1013 Set number of packets to buffer for handling of reordered packets.
1016 Set socket TCP I/O timeout in microseconds.
1019 Override User-Agent header. If not specified, it defaults to the
1020 libavformat identifier string.
1023 When receiving data over UDP, the demuxer tries to reorder received packets
1024 (since they may arrive out of order, or packets may get lost totally). This
1025 can be disabled by setting the maximum demuxing delay to zero (via
1026 the @code{max_delay} field of AVFormatContext).
1028 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1029 streams to display can be chosen with @code{-vst} @var{n} and
1030 @code{-ast} @var{n} for video and audio respectively, and can be switched
1031 on the fly by pressing @code{v} and @code{a}.
1033 @subsection Examples
1035 The following examples all make use of the @command{ffplay} and
1036 @command{ffmpeg} tools.
1040 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1042 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1046 Watch a stream tunneled over HTTP:
1048 ffplay -rtsp_transport http rtsp://server/video.mp4
1052 Send a stream in realtime to a RTSP server, for others to watch:
1054 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1058 Receive a stream in realtime:
1060 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1066 Session Announcement Protocol (RFC 2974). This is not technically a
1067 protocol handler in libavformat, it is a muxer and demuxer.
1068 It is used for signalling of RTP streams, by announcing the SDP for the
1069 streams regularly on a separate port.
1073 The syntax for a SAP url given to the muxer is:
1075 sap://@var{destination}[:@var{port}][?@var{options}]
1078 The RTP packets are sent to @var{destination} on port @var{port},
1079 or to port 5004 if no port is specified.
1080 @var{options} is a @code{&}-separated list. The following options
1085 @item announce_addr=@var{address}
1086 Specify the destination IP address for sending the announcements to.
1087 If omitted, the announcements are sent to the commonly used SAP
1088 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1089 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1091 @item announce_port=@var{port}
1092 Specify the port to send the announcements on, defaults to
1093 9875 if not specified.
1096 Specify the time to live value for the announcements and RTP packets,
1099 @item same_port=@var{0|1}
1100 If set to 1, send all RTP streams on the same port pair. If zero (the
1101 default), all streams are sent on unique ports, with each stream on a
1102 port 2 numbers higher than the previous.
1103 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1104 The RTP stack in libavformat for receiving requires all streams to be sent
1108 Example command lines follow.
1110 To broadcast a stream on the local subnet, for watching in VLC:
1113 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1116 Similarly, for watching in @command{ffplay}:
1119 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1122 And for watching in @command{ffplay}, over IPv6:
1125 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1130 The syntax for a SAP url given to the demuxer is:
1132 sap://[@var{address}][:@var{port}]
1135 @var{address} is the multicast address to listen for announcements on,
1136 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1137 is the port that is listened on, 9875 if omitted.
1139 The demuxers listens for announcements on the given address and port.
1140 Once an announcement is received, it tries to receive that particular stream.
1142 Example command lines follow.
1144 To play back the first stream announced on the normal SAP multicast address:
1150 To play back the first stream announced on one the default IPv6 SAP multicast address:
1153 ffplay sap://[ff0e::2:7ffe]
1158 Stream Control Transmission Protocol.
1160 The accepted URL syntax is:
1162 sctp://@var{host}:@var{port}[?@var{options}]
1165 The protocol accepts the following options:
1168 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1171 Set the maximum number of streams. By default no limit is set.
1176 Haivision Secure Reliable Transport Protocol via libsrt.
1178 The supported syntax for a SRT URL is:
1180 srt://@var{hostname}:@var{port}[?@var{options}]
1183 @var{options} contains a list of &-separated options of the form
1184 @var{key}=@var{val}.
1189 @var{options} srt://@var{hostname}:@var{port}
1192 @var{options} contains a list of '-@var{key} @var{val}'
1195 This protocol accepts the following options.
1198 @item connect_timeout
1199 Connection timeout; SRT cannot connect for RTT > 1500 msec
1200 (2 handshake exchanges) with the default connect timeout of
1201 3 seconds. This option applies to the caller and rendezvous
1202 connection modes. The connect timeout is 10 times the value
1203 set for the rendezvous mode (which can be used as a
1204 workaround for this connection problem with earlier versions).
1206 @item ffs=@var{bytes}
1207 Flight Flag Size (Window Size), in bytes. FFS is actually an
1208 internal parameter and you should set it to not less than
1209 @option{recv_buffer_size} and @option{mss}. The default value
1210 is relatively large, therefore unless you set a very large receiver buffer,
1211 you do not need to change this option. Default value is 25600.
1213 @item inputbw=@var{bytes/seconds}
1214 Sender nominal input rate, in bytes per seconds. Used along with
1215 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1216 calculate maximum sending rate when recovery packets are sent
1217 along with the main media stream:
1218 @option{inputbw} * (100 + @option{oheadbw}) / 100
1219 if @option{inputbw} is not set while @option{maxbw} is set to
1220 relative (0), the actual input rate is evaluated inside
1221 the library. Default value is 0.
1223 @item iptos=@var{tos}
1224 IP Type of Service. Applies to sender only. Default value is 0xB8.
1226 @item ipttl=@var{ttl}
1227 IP Time To Live. Applies to sender only. Default value is 64.
1230 Timestamp-based Packet Delivery Delay.
1231 Used to absorb bursts of missed packet retransmissions.
1232 This flag sets both @option{rcvlatency} and @option{peerlatency}
1233 to the same value. Note that prior to version 1.3.0
1234 this is the only flag to set the latency, however
1235 this is effectively equivalent to setting @option{peerlatency},
1236 when side is sender and @option{rcvlatency}
1237 when side is receiver, and the bidirectional stream
1238 sending is not supported.
1240 @item listen_timeout
1241 Set socket listen timeout.
1243 @item maxbw=@var{bytes/seconds}
1244 Maximum sending bandwidth, in bytes per seconds.
1245 -1 infinite (CSRTCC limit is 30mbps)
1246 0 relative to input rate (see @option{inputbw})
1247 >0 absolute limit value
1248 Default value is 0 (relative)
1250 @item mode=@var{caller|listener|rendezvous}
1252 @option{caller} opens client connection.
1253 @option{listener} starts server to listen for incoming connections.
1254 @option{rendezvous} use Rendez-Vous connection mode.
1255 Default value is caller.
1257 @item mss=@var{bytes}
1258 Maximum Segment Size, in bytes. Used for buffer allocation
1259 and rate calculation using a packet counter assuming fully
1260 filled packets. The smallest MSS between the peers is
1261 used. This is 1500 by default in the overall internet.
1262 This is the maximum size of the UDP packet and can be
1263 only decreased, unless you have some unusual dedicated
1264 network settings. Default value is 1500.
1266 @item nakreport=@var{1|0}
1267 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1268 periodically until a lost packet is retransmitted or
1269 intentionally dropped. Default value is 1.
1271 @item oheadbw=@var{percents}
1272 Recovery bandwidth overhead above input rate, in percents.
1273 See @option{inputbw}. Default value is 25%.
1275 @item passphrase=@var{string}
1276 HaiCrypt Encryption/Decryption Passphrase string, length
1277 from 10 to 79 characters. The passphrase is the shared
1278 secret between the sender and the receiver. It is used
1279 to generate the Key Encrypting Key using PBKDF2
1280 (Password-Based Key Derivation Function). It is used
1281 only if @option{pbkeylen} is non-zero. It is used on
1282 the receiver only if the received data is encrypted.
1283 The configured passphrase cannot be recovered (write-only).
1285 @item enforced_encryption=@var{1|0}
1286 If true, both connection parties must have the same password
1287 set (including empty, that is, with no encryption). If the
1288 password doesn't match or only one side is unencrypted,
1289 the connection is rejected. Default is true.
1291 @item kmrefreshrate=@var{packets}
1292 The number of packets to be transmitted after which the
1293 encryption key is switched to a new key. Default is -1.
1294 -1 means auto (0x1000000 in srt library). The range for
1295 this option is integers in the 0 - @code{INT_MAX}.
1297 @item kmpreannounce=@var{packets}
1298 The interval between when a new encryption key is sent and
1299 when switchover occurs. This value also applies to the
1300 subsequent interval between when switchover occurs and
1301 when the old encryption key is decommissioned. Default is -1.
1302 -1 means auto (0x1000 in srt library). The range for
1303 this option is integers in the 0 - @code{INT_MAX}.
1305 @item payload_size=@var{bytes}
1306 Sets the maximum declared size of a packet transferred
1307 during the single call to the sending function in Live
1308 mode. Use 0 if this value isn't used (which is default in
1310 Default is -1 (automatic), which typically means MPEG-TS;
1311 if you are going to use SRT
1312 to send any different kind of payload, such as, for example,
1313 wrapping a live stream in very small frames, then you can
1314 use a bigger maximum frame size, though not greater than
1317 @item pkt_size=@var{bytes}
1318 Alias for @samp{payload_size}.
1321 The latency value (as described in @option{rcvlatency}) that is
1322 set by the sender side as a minimum value for the receiver.
1324 @item pbkeylen=@var{bytes}
1325 Sender encryption key length, in bytes.
1326 Only can be set to 0, 16, 24 and 32.
1327 Enable sender encryption if not 0.
1328 Not required on receiver (set to 0),
1329 key size obtained from sender in HaiCrypt handshake.
1333 The time that should elapse since the moment when the
1334 packet was sent and the moment when it's delivered to
1335 the receiver application in the receiving function.
1336 This time should be a buffer time large enough to cover
1337 the time spent for sending, unexpectedly extended RTT
1338 time, and the time needed to retransmit the lost UDP
1339 packet. The effective latency value will be the maximum
1340 of this options' value and the value of @option{peerlatency}
1341 set by the peer side. Before version 1.3.0 this option
1342 is only available as @option{latency}.
1344 @item recv_buffer_size=@var{bytes}
1345 Set UDP receive buffer size, expressed in bytes.
1347 @item send_buffer_size=@var{bytes}
1348 Set UDP send buffer size, expressed in bytes.
1351 Set raise error timeout for read/write optations.
1353 This option is only relevant in read mode:
1354 if no data arrived in more than this time
1355 interval, raise error.
1357 @item tlpktdrop=@var{1|0}
1358 Too-late Packet Drop. When enabled on receiver, it skips
1359 missing packets that have not been delivered in time and
1360 delivers the following packets to the application when
1361 their time-to-play has come. It also sends a fake ACK to
1362 the sender. When enabled on sender and enabled on the
1363 receiving peer, the sender drops the older packets that
1364 have no chance of being delivered in time. It was
1365 automatically enabled in the sender if the receiver
1368 @item sndbuf=@var{bytes}
1369 Set send buffer size, expressed in bytes.
1371 @item rcvbuf=@var{bytes}
1372 Set receive buffer size, expressed in bytes.
1374 Receive buffer must not be greater than @option{ffs}.
1376 @item lossmaxttl=@var{packets}
1377 The value up to which the Reorder Tolerance may grow. When
1378 Reorder Tolerance is > 0, then packet loss report is delayed
1379 until that number of packets come in. Reorder Tolerance
1380 increases every time a "belated" packet has come, but it
1381 wasn't due to retransmission (that is, when UDP packets tend
1382 to come out of order), with the difference between the latest
1383 sequence and this packet's sequence, and not more than the
1384 value of this option. By default it's 0, which means that this
1385 mechanism is turned off, and the loss report is always sent
1386 immediately upon experiencing a "gap" in sequences.
1389 The minimum SRT version that is required from the peer. A connection
1390 to a peer that does not satisfy the minimum version requirement
1393 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1396 @item streamid=@var{string}
1397 A string limited to 512 characters that can be set on the socket prior
1398 to connecting. This stream ID will be able to be retrieved by the
1399 listener side from the socket that is returned from srt_accept and
1400 was connected by a socket with that set stream ID. SRT does not enforce
1401 any special interpretation of the contents of this string.
1402 This option doesn’t make sense in Rendezvous connection; the result
1403 might be that simply one side will override the value from the other
1404 side and it’s the matter of luck which one would win
1406 @item smoother=@var{live|file}
1407 The type of Smoother used for the transmission for that socket, which
1408 is responsible for the transmission and congestion control. The Smoother
1409 type must be exactly the same on both connecting parties, otherwise
1410 the connection is rejected.
1412 @item messageapi=@var{1|0}
1413 When set, this socket uses the Message API, otherwise it uses Buffer
1414 API. Note that in live mode (see @option{transtype}) there’s only
1415 message API available. In File mode you can chose to use one of two modes:
1417 Stream API (default, when this option is false). In this mode you may
1418 send as many data as you wish with one sending instruction, or even use
1419 dedicated functions that read directly from a file. The internal facility
1420 will take care of any speed and congestion control. When receiving, you
1421 can also receive as many data as desired, the data not extracted will be
1422 waiting for the next call. There is no boundary between data portions in
1425 Message API. In this mode your single sending instruction passes exactly
1426 one piece of data that has boundaries (a message). Contrary to Live mode,
1427 this message may span across multiple UDP packets and the only size
1428 limitation is that it shall fit as a whole in the sending buffer. The
1429 receiver shall use as large buffer as necessary to receive the message,
1430 otherwise the message will not be given up. When the message is not
1431 complete (not all packets received or there was a packet loss) it will
1434 @item transtype=@var{live|file}
1435 Sets the transmission type for the socket, in particular, setting this
1436 option sets multiple other parameters to their default values as required
1437 for a particular transmission type.
1439 live: Set options as for live transmission. In this mode, you should
1440 send by one sending instruction only so many data that fit in one UDP packet,
1441 and limited to the value defined first in @option{payload_size} (1316 is
1442 default in this mode). There is no speed control in this mode, only the
1443 bandwidth control, if configured, in order to not exceed the bandwidth with
1444 the overhead transmission (retransmitted and control packets).
1446 file: Set options as for non-live transmission. See @option{messageapi}
1447 for further explanations
1449 @item linger=@var{seconds}
1450 The number of seconds that the socket waits for unsent data when closing.
1451 Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
1452 seconds in file mode). The range for this option is integers in the
1457 For more information see: @url{https://github.com/Haivision/srt}.
1461 Secure Real-time Transport Protocol.
1463 The accepted options are:
1466 @item srtp_out_suite
1467 Select input and output encoding suites.
1471 @item AES_CM_128_HMAC_SHA1_80
1472 @item SRTP_AES128_CM_HMAC_SHA1_80
1473 @item AES_CM_128_HMAC_SHA1_32
1474 @item SRTP_AES128_CM_HMAC_SHA1_32
1477 @item srtp_in_params
1478 @item srtp_out_params
1479 Set input and output encoding parameters, which are expressed by a
1480 base64-encoded representation of a binary block. The first 16 bytes of
1481 this binary block are used as master key, the following 14 bytes are
1482 used as master salt.
1487 Virtually extract a segment of a file or another stream.
1488 The underlying stream must be seekable.
1493 Start offset of the extracted segment, in bytes.
1495 End offset of the extracted segment, in bytes.
1496 If set to 0, extract till end of file.
1501 Extract a chapter from a DVD VOB file (start and end sectors obtained
1502 externally and multiplied by 2048):
1504 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1507 Play an AVI file directly from a TAR archive:
1509 subfile,,start,183241728,end,366490624,,:archive.tar
1512 Play a MPEG-TS file from start offset till end:
1514 subfile,,start,32815239,end,0,,:video.ts
1519 Writes the output to multiple protocols. The individual outputs are separated
1523 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1528 Transmission Control Protocol.
1530 The required syntax for a TCP url is:
1532 tcp://@var{hostname}:@var{port}[?@var{options}]
1535 @var{options} contains a list of &-separated options of the form
1536 @var{key}=@var{val}.
1538 The list of supported options follows.
1541 @item listen=@var{1|0}
1542 Listen for an incoming connection. Default value is 0.
1544 @item timeout=@var{microseconds}
1545 Set raise error timeout, expressed in microseconds.
1547 This option is only relevant in read mode: if no data arrived in more
1548 than this time interval, raise error.
1550 @item listen_timeout=@var{milliseconds}
1551 Set listen timeout, expressed in milliseconds.
1553 @item recv_buffer_size=@var{bytes}
1554 Set receive buffer size, expressed bytes.
1556 @item send_buffer_size=@var{bytes}
1557 Set send buffer size, expressed bytes.
1559 @item tcp_nodelay=@var{1|0}
1560 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1562 @item tcp_mss=@var{bytes}
1563 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1566 The following example shows how to setup a listening TCP connection
1567 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1569 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1570 ffplay tcp://@var{hostname}:@var{port}
1575 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1577 The required syntax for a TLS/SSL url is:
1579 tls://@var{hostname}:@var{port}[?@var{options}]
1582 The following parameters can be set via command line options
1583 (or in code via @code{AVOption}s):
1587 @item ca_file, cafile=@var{filename}
1588 A file containing certificate authority (CA) root certificates to treat
1589 as trusted. If the linked TLS library contains a default this might not
1590 need to be specified for verification to work, but not all libraries and
1591 setups have defaults built in.
1592 The file must be in OpenSSL PEM format.
1594 @item tls_verify=@var{1|0}
1595 If enabled, try to verify the peer that we are communicating with.
1596 Note, if using OpenSSL, this currently only makes sure that the
1597 peer certificate is signed by one of the root certificates in the CA
1598 database, but it does not validate that the certificate actually
1599 matches the host name we are trying to connect to. (With other backends,
1600 the host name is validated as well.)
1602 This is disabled by default since it requires a CA database to be
1603 provided by the caller in many cases.
1605 @item cert_file, cert=@var{filename}
1606 A file containing a certificate to use in the handshake with the peer.
1607 (When operating as server, in listen mode, this is more often required
1608 by the peer, while client certificates only are mandated in certain
1611 @item key_file, key=@var{filename}
1612 A file containing the private key for the certificate.
1614 @item listen=@var{1|0}
1615 If enabled, listen for connections on the provided port, and assume
1616 the server role in the handshake instead of the client role.
1620 Example command lines:
1622 To create a TLS/SSL server that serves an input stream.
1625 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1628 To play back a stream from the TLS/SSL server using @command{ffplay}:
1631 ffplay tls://@var{hostname}:@var{port}
1636 User Datagram Protocol.
1638 The required syntax for an UDP URL is:
1640 udp://@var{hostname}:@var{port}[?@var{options}]
1643 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1645 In case threading is enabled on the system, a circular buffer is used
1646 to store the incoming data, which allows one to reduce loss of data due to
1647 UDP socket buffer overruns. The @var{fifo_size} and
1648 @var{overrun_nonfatal} options are related to this buffer.
1650 The list of supported options follows.
1653 @item buffer_size=@var{size}
1654 Set the UDP maximum socket buffer size in bytes. This is used to set either
1655 the receive or send buffer size, depending on what the socket is used for.
1656 Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
1658 @item bitrate=@var{bitrate}
1659 If set to nonzero, the output will have the specified constant bitrate if the
1660 input has enough packets to sustain it.
1662 @item burst_bits=@var{bits}
1663 When using @var{bitrate} this specifies the maximum number of bits in
1666 @item localport=@var{port}
1667 Override the local UDP port to bind with.
1669 @item localaddr=@var{addr}
1670 Local IP address of a network interface used for sending packets or joining
1673 @item pkt_size=@var{size}
1674 Set the size in bytes of UDP packets.
1676 @item reuse=@var{1|0}
1677 Explicitly allow or disallow reusing UDP sockets.
1680 Set the time to live value (for multicast only).
1682 @item connect=@var{1|0}
1683 Initialize the UDP socket with @code{connect()}. In this case, the
1684 destination address can't be changed with ff_udp_set_remote_url later.
1685 If the destination address isn't known at the start, this option can
1686 be specified in ff_udp_set_remote_url, too.
1687 This allows finding out the source address for the packets with getsockname,
1688 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1689 unreachable" is received.
1690 For receiving, this gives the benefit of only receiving packets from
1691 the specified peer address/port.
1693 @item sources=@var{address}[,@var{address}]
1694 Only receive packets sent from the specified addresses. In case of multicast,
1695 also subscribe to multicast traffic coming from these addresses only.
1697 @item block=@var{address}[,@var{address}]
1698 Ignore packets sent from the specified addresses. In case of multicast, also
1699 exclude the source addresses in the multicast subscription.
1701 @item fifo_size=@var{units}
1702 Set the UDP receiving circular buffer size, expressed as a number of
1703 packets with size of 188 bytes. If not specified defaults to 7*4096.
1705 @item overrun_nonfatal=@var{1|0}
1706 Survive in case of UDP receiving circular buffer overrun. Default
1709 @item timeout=@var{microseconds}
1710 Set raise error timeout, expressed in microseconds.
1712 This option is only relevant in read mode: if no data arrived in more
1713 than this time interval, raise error.
1715 @item broadcast=@var{1|0}
1716 Explicitly allow or disallow UDP broadcasting.
1718 Note that broadcasting may not work properly on networks having
1719 a broadcast storm protection.
1722 @subsection Examples
1726 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1728 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1732 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1733 sized UDP packets, using a large input buffer:
1735 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1739 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1741 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1749 The required syntax for a Unix socket URL is:
1752 unix://@var{filepath}
1755 The following parameters can be set via command line options
1756 (or in code via @code{AVOption}s):
1762 Create the Unix socket in listening mode.
1767 ZeroMQ asynchronous messaging using the libzmq library.
1769 This library supports unicast streaming to multiple clients without relying on
1772 The required syntax for streaming or connecting to a stream is:
1774 zmq:tcp://ip-address:port
1778 Create a localhost stream on port 5555:
1780 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1783 Multiple clients may connect to the stream using:
1785 ffplay zmq:tcp://127.0.0.1:5555
1788 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
1789 The server side binds to a port and publishes data. Clients connect to the
1790 server (via IP address/port) and subscribe to the stream. The order in which
1791 the server and client start generally does not matter.
1793 ffmpeg must be compiled with the --enable-libzmq option to support
1796 Options can be set on the @command{ffmpeg}/@command{ffplay} command
1797 line. The following options are supported:
1802 Forces the maximum packet size for sending/receiving data. The default value is
1803 32,768 bytes. On the server side, this sets the maximum size of sent packets
1804 via ZeroMQ. On the clients, it sets an internal buffer size for receiving
1805 packets. Note that pkt_size on the clients should be equal to or greater than
1806 pkt_size on the server. Otherwise the received message may be truncated causing
1812 @c man end PROTOCOLS