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.
200 FTP (File Transfer Protocol).
202 Read from or write to remote resources using FTP protocol.
204 Following syntax is required.
206 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
209 This protocol accepts the following options.
213 Set timeout in microseconds of socket I/O operations used by the underlying low level
214 operation. By default it is set to -1, which means that the timeout is
217 @item ftp-anonymous-password
218 Password used when login as anonymous user. Typically an e-mail address
221 @item ftp-write-seekable
222 Control seekability of connection during encoding. If set to 1 the
223 resource is supposed to be seekable, if set to 0 it is assumed not
224 to be seekable. Default value is 0.
227 NOTE: Protocol can be used as output, but it is recommended to not do
228 it, unless special care is taken (tests, customized server configuration
229 etc.). Different FTP servers behave in different way during seek
230 operation. ff* tools may produce incomplete content due to server limitations.
232 This protocol accepts the following options:
236 If set to 1, the protocol will retry reading at the end of the file, allowing
237 reading files that still are being written. In order for this to terminate,
238 you either need to use the rw_timeout option, or use the interrupt callback
249 Read Apple HTTP Live Streaming compliant segmented stream as
250 a uniform one. The M3U8 playlists describing the segments can be
251 remote HTTP resources or local files, accessed using the standard
253 The nested protocol is declared by specifying
254 "+@var{proto}" after the hls URI scheme name, where @var{proto}
255 is either "file" or "http".
258 hls+http://host/path/to/remote/resource.m3u8
259 hls+file://path/to/local/resource.m3u8
262 Using this protocol is discouraged - the hls demuxer should work
263 just as well (if not, please report the issues) and is more complete.
264 To use the hls demuxer instead, simply use the direct URLs to the
269 HTTP (Hyper Text Transfer Protocol).
271 This protocol accepts the following options:
275 Control seekability of connection. If set to 1 the resource is
276 supposed to be seekable, if set to 0 it is assumed not to be seekable,
277 if set to -1 it will try to autodetect if it is seekable. Default
281 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
284 Set a specific content type for the POST messages or for listen mode.
287 set HTTP proxy to tunnel through e.g. http://example.com:1234
290 Set custom HTTP headers, can override built in default headers. The
291 value must be a string encoding the headers.
293 @item multiple_requests
294 Use persistent connections if set to 1, default is 0.
297 Set custom HTTP post data.
300 Set the Referer header. Include 'Referer: URL' header in HTTP request.
303 Override the User-Agent header. If not specified the protocol will use a
304 string describing the libavformat build. ("Lavf/<version>")
307 This is a deprecated option, you can use user_agent instead it.
310 Set timeout in microseconds of socket I/O operations used by the underlying low level
311 operation. By default it is set to -1, which means that the timeout is
314 @item reconnect_at_eof
315 If set then eof is treated like an error and causes reconnection, this is useful
316 for live / endless streams.
318 @item reconnect_streamed
319 If set then even streamed/non seekable streams will be reconnected on errors.
321 @item reconnect_delay_max
322 Sets the maximum delay in seconds after which to give up reconnecting
325 Export the MIME type.
328 Exports the HTTP response version number. Usually "1.0" or "1.1".
331 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
332 supports this, the metadata has to be retrieved by the application by reading
333 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
336 @item icy_metadata_headers
337 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
338 headers, separated by newline characters.
340 @item icy_metadata_packet
341 If the server supports ICY metadata, and @option{icy} was set to 1, this
342 contains the last non-empty metadata packet sent by the server. It should be
343 polled in regular intervals by applications interested in mid-stream metadata
347 Set the cookies to be sent in future requests. The format of each cookie is the
348 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
349 delimited by a newline character.
352 Set initial byte offset.
355 Try to limit the request to bytes preceding this offset.
358 When used as a client option it sets the HTTP method for the request.
360 When used as a server option it sets the HTTP method that is going to be
361 expected from the client(s).
362 If the expected and the received HTTP method do not match the client will
363 be given a Bad Request response.
364 When unset the HTTP method is not checked for now. This will be replaced by
365 autodetection in the future.
368 If set to 1 enables experimental HTTP server. This can be used to send data when
369 used as an output option, or read data from a client with HTTP POST when used as
371 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
372 in ffmpeg.c and thus must not be used as a command line option.
374 # Server side (sending):
375 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
377 # Client side (receiving):
378 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
380 # Client can also be done with wget:
381 wget http://@var{server}:@var{port} -O somefile.ogg
383 # Server side (receiving):
384 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
386 # Client side (sending):
387 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
389 # Client can also be done with wget:
390 wget --post-file=somefile.ogg http://@var{server}:@var{port}
395 @subsection HTTP Cookies
397 Some HTTP requests will be denied unless cookie values are passed in with the
398 request. The @option{cookies} option allows these cookies to be specified. At
399 the very least, each cookie must specify a value along with a path and domain.
400 HTTP requests that match both the domain and path will automatically include the
401 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
404 The required syntax to play a stream specifying a cookie is:
406 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
411 Icecast protocol (stream to Icecast servers)
413 This protocol accepts the following options:
417 Set the stream genre.
422 @item ice_description
423 Set the stream description.
426 Set the stream website URL.
429 Set if the stream should be public.
430 The default is 0 (not public).
433 Override the User-Agent header. If not specified a string of the form
434 "Lavf/<version>" will be used.
437 Set the Icecast mountpoint password.
440 Set the stream content type. This must be set if it is different from
444 This enables support for Icecast versions < 2.4.0, that do not support the
445 HTTP PUT method but the SOURCE method.
450 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
455 MMS (Microsoft Media Server) protocol over TCP.
459 MMS (Microsoft Media Server) protocol over HTTP.
461 The required syntax is:
463 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
470 Computes the MD5 hash of the data to be written, and on close writes
471 this to the designated output or stdout if none is specified. It can
472 be used to test muxers without writing an actual file.
474 Some examples follow.
476 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
477 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
479 # Write the MD5 hash of the encoded AVI file to stdout.
480 ffmpeg -i input.flv -f avi -y md5:
483 Note that some formats (typically MOV) require the output protocol to
484 be seekable, so they will fail with the MD5 output protocol.
488 UNIX pipe access protocol.
490 Read and write from UNIX pipes.
492 The accepted syntax is:
497 @var{number} is the number corresponding to the file descriptor of the
498 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
499 is not specified, by default the stdout file descriptor will be used
500 for writing, stdin for reading.
502 For example to read from stdin with @command{ffmpeg}:
504 cat test.wav | ffmpeg -i pipe:0
505 # ...this is the same as...
506 cat test.wav | ffmpeg -i pipe:
509 For writing to stdout with @command{ffmpeg}:
511 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
512 # ...this is the same as...
513 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
516 This protocol accepts the following options:
520 Set I/O operation maximum block size, in bytes. Default value is
521 @code{INT_MAX}, which results in not limiting the requested block size.
522 Setting this value reasonably low improves user termination request reaction
523 time, which is valuable if data transmission is slow.
526 Note that some formats (typically MOV), require the output protocol to
527 be seekable, so they will fail with the pipe output protocol.
531 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
533 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
534 for MPEG-2 Transport Streams sent over RTP.
536 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
537 the @code{rtp} protocol.
539 The required syntax is:
541 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
544 The destination UDP ports are @code{port + 2} for the column FEC stream
545 and @code{port + 4} for the row FEC stream.
547 This protocol accepts the following options:
551 The number of columns (4-20, LxD <= 100)
554 The number of rows (4-20, LxD <= 100)
561 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
566 Real-Time Messaging Protocol.
568 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
569 content across a TCP/IP network.
571 The required syntax is:
573 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
576 The accepted parameters are:
580 An optional username (mostly for publishing).
583 An optional password (mostly for publishing).
586 The address of the RTMP server.
589 The number of the TCP port to use (by default is 1935).
592 It is the name of the application to access. It usually corresponds to
593 the path where the application is installed on the RTMP server
594 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
595 the value parsed from the URI through the @code{rtmp_app} option, too.
598 It is the path or name of the resource to play with reference to the
599 application specified in @var{app}, may be prefixed by "mp4:". You
600 can override the value parsed from the URI through the @code{rtmp_playpath}
604 Act as a server, listening for an incoming connection.
607 Maximum time to wait for the incoming connection. Implies listen.
610 Additionally, the following parameters can be set via command line options
611 (or in code via @code{AVOption}s):
615 Name of application to connect on the RTMP server. This option
616 overrides the parameter specified in the URI.
619 Set the client buffer time in milliseconds. The default is 3000.
622 Extra arbitrary AMF connection parameters, parsed from a string,
623 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
624 Each value is prefixed by a single character denoting the type,
625 B for Boolean, N for number, S for string, O for object, or Z for null,
626 followed by a colon. For Booleans the data must be either 0 or 1 for
627 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
628 1 to end or begin an object, respectively. Data items in subobjects may
629 be named, by prefixing the type with 'N' and specifying the name before
630 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
631 times to construct arbitrary AMF sequences.
634 Version of the Flash plugin used to run the SWF player. The default
635 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
636 <libavformat version>).)
638 @item rtmp_flush_interval
639 Number of packets flushed in the same request (RTMPT only). The default
643 Specify that the media is a live stream. No resuming or seeking in
644 live streams is possible. The default value is @code{any}, which means the
645 subscriber first tries to play the live stream specified in the
646 playpath. If a live stream of that name is not found, it plays the
647 recorded stream. The other possible values are @code{live} and
651 URL of the web page in which the media was embedded. By default no
655 Stream identifier to play or to publish. This option overrides the
656 parameter specified in the URI.
659 Name of live stream to subscribe to. By default no value will be sent.
660 It is only sent if the option is specified or if rtmp_live
664 SHA256 hash of the decompressed SWF file (32 bytes).
667 Size of the decompressed SWF file, required for SWFVerification.
670 URL of the SWF player for the media. By default no value will be sent.
673 URL to player swf file, compute hash/size automatically.
676 URL of the target stream. Defaults to proto://host[:port]/app.
680 For example to read with @command{ffplay} a multimedia resource named
681 "sample" from the application "vod" from an RTMP server "myserver":
683 ffplay rtmp://myserver/vod/sample
686 To publish to a password protected server, passing the playpath and
687 app names separately:
689 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
694 Encrypted Real-Time Messaging Protocol.
696 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
697 streaming multimedia content within standard cryptographic primitives,
698 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
703 Real-Time Messaging Protocol over a secure SSL connection.
705 The Real-Time Messaging Protocol (RTMPS) is used for streaming
706 multimedia content across an encrypted connection.
710 Real-Time Messaging Protocol tunneled through HTTP.
712 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
713 for streaming multimedia content within HTTP requests to traverse
718 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
720 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
721 is used for streaming multimedia content within HTTP requests to traverse
726 Real-Time Messaging Protocol tunneled through HTTPS.
728 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
729 for streaming multimedia content within HTTPS requests to traverse
732 @section libsmbclient
734 libsmbclient permits one to manipulate CIFS/SMB network resources.
736 Following syntax is required.
739 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
742 This protocol accepts the following options.
746 Set timeout in milliseconds of socket I/O operations used by the underlying
747 low level operation. By default it is set to -1, which means that the timeout
751 Truncate existing files on write, if set to 1. A value of 0 prevents
752 truncating. Default value is 1.
755 Set the workgroup used for making connections. By default workgroup is not specified.
759 For more information see: @url{http://www.samba.org/}.
763 Secure File Transfer Protocol via libssh
765 Read from or write to remote resources using SFTP protocol.
767 Following syntax is required.
770 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
773 This protocol accepts the following options.
777 Set timeout of socket I/O operations used by the underlying low level
778 operation. By default it is set to -1, which means that the timeout
782 Truncate existing files on write, if set to 1. A value of 0 prevents
783 truncating. Default value is 1.
786 Specify the path of the file containing private key to use during authorization.
787 By default libssh searches for keys in the @file{~/.ssh/} directory.
791 Example: Play a file stored on remote server.
794 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
797 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
799 Real-Time Messaging Protocol and its variants supported through
802 Requires the presence of the librtmp headers and library during
803 configuration. You need to explicitly configure the build with
804 "--enable-librtmp". If enabled this will replace the native RTMP
807 This protocol provides most client functions and a few server
808 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
809 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
810 variants of these encrypted types (RTMPTE, RTMPTS).
812 The required syntax is:
814 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
817 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
818 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
819 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
820 meaning as specified for the RTMP native protocol.
821 @var{options} contains a list of space-separated options of the form
824 See the librtmp manual page (man 3 librtmp) for more information.
826 For example, to stream a file in real-time to an RTMP server using
829 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
832 To play the same stream using @command{ffplay}:
834 ffplay "rtmp://myserver/live/mystream live=1"
839 Real-time Transport Protocol.
841 The required syntax for an RTP URL is:
842 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
844 @var{port} specifies the RTP port to use.
846 The following URL options are supported:
851 Set the TTL (Time-To-Live) value (for multicast only).
853 @item rtcpport=@var{n}
854 Set the remote RTCP port to @var{n}.
856 @item localrtpport=@var{n}
857 Set the local RTP port to @var{n}.
859 @item localrtcpport=@var{n}'
860 Set the local RTCP port to @var{n}.
862 @item pkt_size=@var{n}
863 Set max packet size (in bytes) to @var{n}.
866 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
869 @item sources=@var{ip}[,@var{ip}]
870 List allowed source IP addresses.
872 @item block=@var{ip}[,@var{ip}]
873 List disallowed (blocked) source IP addresses.
875 @item write_to_source=0|1
876 Send packets to the source address of the latest received packet (if
877 set to 1) or to a default remote address (if set to 0).
879 @item localport=@var{n}
880 Set the local RTP port to @var{n}.
882 This is a deprecated option. Instead, @option{localrtpport} should be
892 If @option{rtcpport} is not set the RTCP port will be set to the RTP
896 If @option{localrtpport} (the local RTP port) is not set any available
897 port will be used for the local RTP and RTCP ports.
900 If @option{localrtcpport} (the local RTCP port) is not set it will be
901 set to the local RTP port value plus 1.
906 Real-Time Streaming Protocol.
908 RTSP is not technically a protocol handler in libavformat, it is a demuxer
909 and muxer. The demuxer supports both normal RTSP (with data transferred
910 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
911 data transferred over RDT).
913 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
914 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
915 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
917 The required syntax for a RTSP url is:
919 rtsp://@var{hostname}[:@var{port}]/@var{path}
922 Options can be set on the @command{ffmpeg}/@command{ffplay} command
923 line, or set in code via @code{AVOption}s or in
924 @code{avformat_open_input}.
926 The following options are supported.
930 Do not start playing the stream immediately if set to 1. Default value
934 Set RTSP transport protocols.
936 It accepts the following values:
939 Use UDP as lower transport protocol.
942 Use TCP (interleaving within the RTSP control channel) as lower
946 Use UDP multicast as lower transport protocol.
949 Use HTTP tunneling as lower transport protocol, which is useful for
953 Multiple lower transport protocols may be specified, in that case they are
954 tried one at a time (if the setup of one fails, the next one is tried).
955 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
960 The following values are accepted:
963 Accept packets only from negotiated peer address and port.
965 Act as a server, listening for an incoming connection.
967 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
970 Default value is @samp{none}.
972 @item allowed_media_types
973 Set media types to accept from the server.
975 The following flags are accepted:
982 By default it accepts all media types.
985 Set minimum local UDP port. Default value is 5000.
988 Set maximum local UDP port. Default value is 65000.
991 Set maximum timeout (in seconds) to wait for incoming connections.
993 A value of -1 means infinite (default). This option implies the
994 @option{rtsp_flags} set to @samp{listen}.
996 @item reorder_queue_size
997 Set number of packets to buffer for handling of reordered packets.
1000 Set socket TCP I/O timeout in microseconds.
1003 Override User-Agent header. If not specified, it defaults to the
1004 libavformat identifier string.
1007 When receiving data over UDP, the demuxer tries to reorder received packets
1008 (since they may arrive out of order, or packets may get lost totally). This
1009 can be disabled by setting the maximum demuxing delay to zero (via
1010 the @code{max_delay} field of AVFormatContext).
1012 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1013 streams to display can be chosen with @code{-vst} @var{n} and
1014 @code{-ast} @var{n} for video and audio respectively, and can be switched
1015 on the fly by pressing @code{v} and @code{a}.
1017 @subsection Examples
1019 The following examples all make use of the @command{ffplay} and
1020 @command{ffmpeg} tools.
1024 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1026 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1030 Watch a stream tunneled over HTTP:
1032 ffplay -rtsp_transport http rtsp://server/video.mp4
1036 Send a stream in realtime to a RTSP server, for others to watch:
1038 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1042 Receive a stream in realtime:
1044 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1050 Session Announcement Protocol (RFC 2974). This is not technically a
1051 protocol handler in libavformat, it is a muxer and demuxer.
1052 It is used for signalling of RTP streams, by announcing the SDP for the
1053 streams regularly on a separate port.
1057 The syntax for a SAP url given to the muxer is:
1059 sap://@var{destination}[:@var{port}][?@var{options}]
1062 The RTP packets are sent to @var{destination} on port @var{port},
1063 or to port 5004 if no port is specified.
1064 @var{options} is a @code{&}-separated list. The following options
1069 @item announce_addr=@var{address}
1070 Specify the destination IP address for sending the announcements to.
1071 If omitted, the announcements are sent to the commonly used SAP
1072 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1073 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1075 @item announce_port=@var{port}
1076 Specify the port to send the announcements on, defaults to
1077 9875 if not specified.
1080 Specify the time to live value for the announcements and RTP packets,
1083 @item same_port=@var{0|1}
1084 If set to 1, send all RTP streams on the same port pair. If zero (the
1085 default), all streams are sent on unique ports, with each stream on a
1086 port 2 numbers higher than the previous.
1087 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1088 The RTP stack in libavformat for receiving requires all streams to be sent
1092 Example command lines follow.
1094 To broadcast a stream on the local subnet, for watching in VLC:
1097 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1100 Similarly, for watching in @command{ffplay}:
1103 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1106 And for watching in @command{ffplay}, over IPv6:
1109 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1114 The syntax for a SAP url given to the demuxer is:
1116 sap://[@var{address}][:@var{port}]
1119 @var{address} is the multicast address to listen for announcements on,
1120 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1121 is the port that is listened on, 9875 if omitted.
1123 The demuxers listens for announcements on the given address and port.
1124 Once an announcement is received, it tries to receive that particular stream.
1126 Example command lines follow.
1128 To play back the first stream announced on the normal SAP multicast address:
1134 To play back the first stream announced on one the default IPv6 SAP multicast address:
1137 ffplay sap://[ff0e::2:7ffe]
1142 Stream Control Transmission Protocol.
1144 The accepted URL syntax is:
1146 sctp://@var{host}:@var{port}[?@var{options}]
1149 The protocol accepts the following options:
1152 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1155 Set the maximum number of streams. By default no limit is set.
1160 Haivision Secure Reliable Transport Protocol via libsrt.
1162 The supported syntax for a SRT URL is:
1164 srt://@var{hostname}:@var{port}[?@var{options}]
1167 @var{options} contains a list of &-separated options of the form
1168 @var{key}=@var{val}.
1173 @var{options} srt://@var{hostname}:@var{port}
1176 @var{options} contains a list of '-@var{key} @var{val}'
1179 This protocol accepts the following options.
1182 @item connect_timeout
1183 Connection timeout; SRT cannot connect for RTT > 1500 msec
1184 (2 handshake exchanges) with the default connect timeout of
1185 3 seconds. This option applies to the caller and rendezvous
1186 connection modes. The connect timeout is 10 times the value
1187 set for the rendezvous mode (which can be used as a
1188 workaround for this connection problem with earlier versions).
1190 @item ffs=@var{bytes}
1191 Flight Flag Size (Window Size), in bytes. FFS is actually an
1192 internal parameter and you should set it to not less than
1193 @option{recv_buffer_size} and @option{mss}. The default value
1194 is relatively large, therefore unless you set a very large receiver buffer,
1195 you do not need to change this option. Default value is 25600.
1197 @item inputbw=@var{bytes/seconds}
1198 Sender nominal input rate, in bytes per seconds. Used along with
1199 @option{oheadbw}, when @option{maxbw} is set to relative (0), to
1200 calculate maximum sending rate when recovery packets are sent
1201 along with the main media stream:
1202 @option{inputbw} * (100 + @option{oheadbw}) / 100
1203 if @option{inputbw} is not set while @option{maxbw} is set to
1204 relative (0), the actual input rate is evaluated inside
1205 the library. Default value is 0.
1207 @item iptos=@var{tos}
1208 IP Type of Service. Applies to sender only. Default value is 0xB8.
1210 @item ipttl=@var{ttl}
1211 IP Time To Live. Applies to sender only. Default value is 64.
1214 Timestamp-based Packet Delivery Delay.
1215 Used to absorb bursts of missed packet retransmissions.
1216 This flag sets both @option{rcvlatency} and @option{peerlatency}
1217 to the same value. Note that prior to version 1.3.0
1218 this is the only flag to set the latency, however
1219 this is effectively equivalent to setting @option{peerlatency},
1220 when side is sender and @option{rcvlatency}
1221 when side is receiver, and the bidirectional stream
1222 sending is not supported.
1224 @item listen_timeout
1225 Set socket listen timeout.
1227 @item maxbw=@var{bytes/seconds}
1228 Maximum sending bandwidth, in bytes per seconds.
1229 -1 infinite (CSRTCC limit is 30mbps)
1230 0 relative to input rate (see @option{inputbw})
1231 >0 absolute limit value
1232 Default value is 0 (relative)
1234 @item mode=@var{caller|listener|rendezvous}
1236 @option{caller} opens client connection.
1237 @option{listener} starts server to listen for incoming connections.
1238 @option{rendezvous} use Rendez-Vous connection mode.
1239 Default value is caller.
1241 @item mss=@var{bytes}
1242 Maximum Segment Size, in bytes. Used for buffer allocation
1243 and rate calculation using a packet counter assuming fully
1244 filled packets. The smallest MSS between the peers is
1245 used. This is 1500 by default in the overall internet.
1246 This is the maximum size of the UDP packet and can be
1247 only decreased, unless you have some unusual dedicated
1248 network settings. Default value is 1500.
1250 @item nakreport=@var{1|0}
1251 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1252 periodically until a lost packet is retransmitted or
1253 intentionally dropped. Default value is 1.
1255 @item oheadbw=@var{percents}
1256 Recovery bandwidth overhead above input rate, in percents.
1257 See @option{inputbw}. Default value is 25%.
1259 @item passphrase=@var{string}
1260 HaiCrypt Encryption/Decryption Passphrase string, length
1261 from 10 to 79 characters. The passphrase is the shared
1262 secret between the sender and the receiver. It is used
1263 to generate the Key Encrypting Key using PBKDF2
1264 (Password-Based Key Derivation Function). It is used
1265 only if @option{pbkeylen} is non-zero. It is used on
1266 the receiver only if the received data is encrypted.
1267 The configured passphrase cannot be recovered (write-only).
1269 @item payload_size=@var{bytes}
1270 Sets the maximum declared size of a packet transferred
1271 during the single call to the sending function in Live
1272 mode. Use 0 if this value isn't used (which is default in
1274 Default is -1 (automatic), which typically means MPEG-TS;
1275 if you are going to use SRT
1276 to send any different kind of payload, such as, for example,
1277 wrapping a live stream in very small frames, then you can
1278 use a bigger maximum frame size, though not greater than
1281 @item pkt_size=@var{bytes}
1282 Alias for @samp{payload_size}.
1285 The latency value (as described in @option{rcvlatency}) that is
1286 set by the sender side as a minimum value for the receiver.
1288 @item pbkeylen=@var{bytes}
1289 Sender encryption key length, in bytes.
1290 Only can be set to 0, 16, 24 and 32.
1291 Enable sender encryption if not 0.
1292 Not required on receiver (set to 0),
1293 key size obtained from sender in HaiCrypt handshake.
1297 The time that should elapse since the moment when the
1298 packet was sent and the moment when it's delivered to
1299 the receiver application in the receiving function.
1300 This time should be a buffer time large enough to cover
1301 the time spent for sending, unexpectedly extended RTT
1302 time, and the time needed to retransmit the lost UDP
1303 packet. The effective latency value will be the maximum
1304 of this options' value and the value of @option{peerlatency}
1305 set by the peer side. Before version 1.3.0 this option
1306 is only available as @option{latency}.
1308 @item recv_buffer_size=@var{bytes}
1309 Set receive buffer size, expressed in bytes.
1311 @item send_buffer_size=@var{bytes}
1312 Set send buffer size, expressed in bytes.
1315 Set raise error timeout for read/write optations.
1317 This option is only relevant in read mode:
1318 if no data arrived in more than this time
1319 interval, raise error.
1321 @item tlpktdrop=@var{1|0}
1322 Too-late Packet Drop. When enabled on receiver, it skips
1323 missing packets that have not been delivered in time and
1324 delivers the following packets to the application when
1325 their time-to-play has come. It also sends a fake ACK to
1326 the sender. When enabled on sender and enabled on the
1327 receiving peer, the sender drops the older packets that
1328 have no chance of being delivered in time. It was
1329 automatically enabled in the sender if the receiver
1334 For more information see: @url{https://github.com/Haivision/srt}.
1338 Secure Real-time Transport Protocol.
1340 The accepted options are:
1343 @item srtp_out_suite
1344 Select input and output encoding suites.
1348 @item AES_CM_128_HMAC_SHA1_80
1349 @item SRTP_AES128_CM_HMAC_SHA1_80
1350 @item AES_CM_128_HMAC_SHA1_32
1351 @item SRTP_AES128_CM_HMAC_SHA1_32
1354 @item srtp_in_params
1355 @item srtp_out_params
1356 Set input and output encoding parameters, which are expressed by a
1357 base64-encoded representation of a binary block. The first 16 bytes of
1358 this binary block are used as master key, the following 14 bytes are
1359 used as master salt.
1364 Virtually extract a segment of a file or another stream.
1365 The underlying stream must be seekable.
1370 Start offset of the extracted segment, in bytes.
1372 End offset of the extracted segment, in bytes.
1373 If set to 0, extract till end of file.
1378 Extract a chapter from a DVD VOB file (start and end sectors obtained
1379 externally and multiplied by 2048):
1381 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1384 Play an AVI file directly from a TAR archive:
1386 subfile,,start,183241728,end,366490624,,:archive.tar
1389 Play a MPEG-TS file from start offset till end:
1391 subfile,,start,32815239,end,0,,:video.ts
1396 Writes the output to multiple protocols. The individual outputs are separated
1400 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1405 Transmission Control Protocol.
1407 The required syntax for a TCP url is:
1409 tcp://@var{hostname}:@var{port}[?@var{options}]
1412 @var{options} contains a list of &-separated options of the form
1413 @var{key}=@var{val}.
1415 The list of supported options follows.
1418 @item listen=@var{1|0}
1419 Listen for an incoming connection. Default value is 0.
1421 @item timeout=@var{microseconds}
1422 Set raise error timeout, expressed in microseconds.
1424 This option is only relevant in read mode: if no data arrived in more
1425 than this time interval, raise error.
1427 @item listen_timeout=@var{milliseconds}
1428 Set listen timeout, expressed in milliseconds.
1430 @item recv_buffer_size=@var{bytes}
1431 Set receive buffer size, expressed bytes.
1433 @item send_buffer_size=@var{bytes}
1434 Set send buffer size, expressed bytes.
1436 @item tcp_nodelay=@var{1|0}
1437 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1439 @item tcp_mss=@var{bytes}
1440 Set maximum segment size for outgoing TCP packets, expressed in bytes.
1443 The following example shows how to setup a listening TCP connection
1444 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1446 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1447 ffplay tcp://@var{hostname}:@var{port}
1452 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1454 The required syntax for a TLS/SSL url is:
1456 tls://@var{hostname}:@var{port}[?@var{options}]
1459 The following parameters can be set via command line options
1460 (or in code via @code{AVOption}s):
1464 @item ca_file, cafile=@var{filename}
1465 A file containing certificate authority (CA) root certificates to treat
1466 as trusted. If the linked TLS library contains a default this might not
1467 need to be specified for verification to work, but not all libraries and
1468 setups have defaults built in.
1469 The file must be in OpenSSL PEM format.
1471 @item tls_verify=@var{1|0}
1472 If enabled, try to verify the peer that we are communicating with.
1473 Note, if using OpenSSL, this currently only makes sure that the
1474 peer certificate is signed by one of the root certificates in the CA
1475 database, but it does not validate that the certificate actually
1476 matches the host name we are trying to connect to. (With other backends,
1477 the host name is validated as well.)
1479 This is disabled by default since it requires a CA database to be
1480 provided by the caller in many cases.
1482 @item cert_file, cert=@var{filename}
1483 A file containing a certificate to use in the handshake with the peer.
1484 (When operating as server, in listen mode, this is more often required
1485 by the peer, while client certificates only are mandated in certain
1488 @item key_file, key=@var{filename}
1489 A file containing the private key for the certificate.
1491 @item listen=@var{1|0}
1492 If enabled, listen for connections on the provided port, and assume
1493 the server role in the handshake instead of the client role.
1497 Example command lines:
1499 To create a TLS/SSL server that serves an input stream.
1502 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1505 To play back a stream from the TLS/SSL server using @command{ffplay}:
1508 ffplay tls://@var{hostname}:@var{port}
1513 User Datagram Protocol.
1515 The required syntax for an UDP URL is:
1517 udp://@var{hostname}:@var{port}[?@var{options}]
1520 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1522 In case threading is enabled on the system, a circular buffer is used
1523 to store the incoming data, which allows one to reduce loss of data due to
1524 UDP socket buffer overruns. The @var{fifo_size} and
1525 @var{overrun_nonfatal} options are related to this buffer.
1527 The list of supported options follows.
1530 @item buffer_size=@var{size}
1531 Set the UDP maximum socket buffer size in bytes. This is used to set either
1532 the receive or send buffer size, depending on what the socket is used for.
1533 Default is 64KB. See also @var{fifo_size}.
1535 @item bitrate=@var{bitrate}
1536 If set to nonzero, the output will have the specified constant bitrate if the
1537 input has enough packets to sustain it.
1539 @item burst_bits=@var{bits}
1540 When using @var{bitrate} this specifies the maximum number of bits in
1543 @item localport=@var{port}
1544 Override the local UDP port to bind with.
1546 @item localaddr=@var{addr}
1547 Choose the local IP address. This is useful e.g. if sending multicast
1548 and the host has multiple interfaces, where the user can choose
1549 which interface to send on by specifying the IP address of that interface.
1551 @item pkt_size=@var{size}
1552 Set the size in bytes of UDP packets.
1554 @item reuse=@var{1|0}
1555 Explicitly allow or disallow reusing UDP sockets.
1558 Set the time to live value (for multicast only).
1560 @item connect=@var{1|0}
1561 Initialize the UDP socket with @code{connect()}. In this case, the
1562 destination address can't be changed with ff_udp_set_remote_url later.
1563 If the destination address isn't known at the start, this option can
1564 be specified in ff_udp_set_remote_url, too.
1565 This allows finding out the source address for the packets with getsockname,
1566 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1567 unreachable" is received.
1568 For receiving, this gives the benefit of only receiving packets from
1569 the specified peer address/port.
1571 @item sources=@var{address}[,@var{address}]
1572 Only receive packets sent to the multicast group from one of the
1573 specified sender IP addresses.
1575 @item block=@var{address}[,@var{address}]
1576 Ignore packets sent to the multicast group from the specified
1577 sender IP addresses.
1579 @item fifo_size=@var{units}
1580 Set the UDP receiving circular buffer size, expressed as a number of
1581 packets with size of 188 bytes. If not specified defaults to 7*4096.
1583 @item overrun_nonfatal=@var{1|0}
1584 Survive in case of UDP receiving circular buffer overrun. Default
1587 @item timeout=@var{microseconds}
1588 Set raise error timeout, expressed in microseconds.
1590 This option is only relevant in read mode: if no data arrived in more
1591 than this time interval, raise error.
1593 @item broadcast=@var{1|0}
1594 Explicitly allow or disallow UDP broadcasting.
1596 Note that broadcasting may not work properly on networks having
1597 a broadcast storm protection.
1600 @subsection Examples
1604 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1606 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1610 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1611 sized UDP packets, using a large input buffer:
1613 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1617 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1619 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1627 The required syntax for a Unix socket URL is:
1630 unix://@var{filepath}
1633 The following parameters can be set via command line options
1634 (or in code via @code{AVOption}s):
1640 Create the Unix socket in listening mode.
1643 @c man end PROTOCOLS