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 Override the User-Agent header. If not specified the protocol will use a
301 string describing the libavformat build. ("Lavf/<version>")
304 This is a deprecated option, you can use user_agent instead it.
307 Set timeout in microseconds of socket I/O operations used by the underlying low level
308 operation. By default it is set to -1, which means that the timeout is
311 @item reconnect_at_eof
312 If set then eof is treated like an error and causes reconnection, this is useful
313 for live / endless streams.
315 @item reconnect_streamed
316 If set then even streamed/non seekable streams will be reconnected on errors.
318 @item reconnect_delay_max
319 Sets the maximum delay in seconds after which to give up reconnecting
322 Export the MIME type.
325 Exports the HTTP response version number. Usually "1.0" or "1.1".
328 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
329 supports this, the metadata has to be retrieved by the application by reading
330 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
333 @item icy_metadata_headers
334 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
335 headers, separated by newline characters.
337 @item icy_metadata_packet
338 If the server supports ICY metadata, and @option{icy} was set to 1, this
339 contains the last non-empty metadata packet sent by the server. It should be
340 polled in regular intervals by applications interested in mid-stream metadata
344 Set the cookies to be sent in future requests. The format of each cookie is the
345 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
346 delimited by a newline character.
349 Set initial byte offset.
352 Try to limit the request to bytes preceding this offset.
355 When used as a client option it sets the HTTP method for the request.
357 When used as a server option it sets the HTTP method that is going to be
358 expected from the client(s).
359 If the expected and the received HTTP method do not match the client will
360 be given a Bad Request response.
361 When unset the HTTP method is not checked for now. This will be replaced by
362 autodetection in the future.
365 If set to 1 enables experimental HTTP server. This can be used to send data when
366 used as an output option, or read data from a client with HTTP POST when used as
368 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
369 in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
371 # Server side (sending):
372 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
374 # Client side (receiving):
375 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
377 # Client can also be done with wget:
378 wget http://@var{server}:@var{port} -O somefile.ogg
380 # Server side (receiving):
381 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
383 # Client side (sending):
384 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
386 # Client can also be done with wget:
387 wget --post-file=somefile.ogg http://@var{server}:@var{port}
392 @subsection HTTP Cookies
394 Some HTTP requests will be denied unless cookie values are passed in with the
395 request. The @option{cookies} option allows these cookies to be specified. At
396 the very least, each cookie must specify a value along with a path and domain.
397 HTTP requests that match both the domain and path will automatically include the
398 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
401 The required syntax to play a stream specifying a cookie is:
403 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
408 Icecast protocol (stream to Icecast servers)
410 This protocol accepts the following options:
414 Set the stream genre.
419 @item ice_description
420 Set the stream description.
423 Set the stream website URL.
426 Set if the stream should be public.
427 The default is 0 (not public).
430 Override the User-Agent header. If not specified a string of the form
431 "Lavf/<version>" will be used.
434 Set the Icecast mountpoint password.
437 Set the stream content type. This must be set if it is different from
441 This enables support for Icecast versions < 2.4.0, that do not support the
442 HTTP PUT method but the SOURCE method.
447 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
452 MMS (Microsoft Media Server) protocol over TCP.
456 MMS (Microsoft Media Server) protocol over HTTP.
458 The required syntax is:
460 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
467 Computes the MD5 hash of the data to be written, and on close writes
468 this to the designated output or stdout if none is specified. It can
469 be used to test muxers without writing an actual file.
471 Some examples follow.
473 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
474 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
476 # Write the MD5 hash of the encoded AVI file to stdout.
477 ffmpeg -i input.flv -f avi -y md5:
480 Note that some formats (typically MOV) require the output protocol to
481 be seekable, so they will fail with the MD5 output protocol.
485 UNIX pipe access protocol.
487 Read and write from UNIX pipes.
489 The accepted syntax is:
494 @var{number} is the number corresponding to the file descriptor of the
495 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
496 is not specified, by default the stdout file descriptor will be used
497 for writing, stdin for reading.
499 For example to read from stdin with @command{ffmpeg}:
501 cat test.wav | ffmpeg -i pipe:0
502 # ...this is the same as...
503 cat test.wav | ffmpeg -i pipe:
506 For writing to stdout with @command{ffmpeg}:
508 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
509 # ...this is the same as...
510 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
513 This protocol accepts the following options:
517 Set I/O operation maximum block size, in bytes. Default value is
518 @code{INT_MAX}, which results in not limiting the requested block size.
519 Setting this value reasonably low improves user termination request reaction
520 time, which is valuable if data transmission is slow.
523 Note that some formats (typically MOV), require the output protocol to
524 be seekable, so they will fail with the pipe output protocol.
528 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
530 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
531 for MPEG-2 Transport Streams sent over RTP.
533 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
534 the @code{rtp} protocol.
536 The required syntax is:
538 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
541 The destination UDP ports are @code{port + 2} for the column FEC stream
542 and @code{port + 4} for the row FEC stream.
544 This protocol accepts the following options:
548 The number of columns (4-20, LxD <= 100)
551 The number of rows (4-20, LxD <= 100)
558 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
563 Real-Time Messaging Protocol.
565 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
566 content across a TCP/IP network.
568 The required syntax is:
570 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
573 The accepted parameters are:
577 An optional username (mostly for publishing).
580 An optional password (mostly for publishing).
583 The address of the RTMP server.
586 The number of the TCP port to use (by default is 1935).
589 It is the name of the application to access. It usually corresponds to
590 the path where the application is installed on the RTMP server
591 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
592 the value parsed from the URI through the @code{rtmp_app} option, too.
595 It is the path or name of the resource to play with reference to the
596 application specified in @var{app}, may be prefixed by "mp4:". You
597 can override the value parsed from the URI through the @code{rtmp_playpath}
601 Act as a server, listening for an incoming connection.
604 Maximum time to wait for the incoming connection. Implies listen.
607 Additionally, the following parameters can be set via command line options
608 (or in code via @code{AVOption}s):
612 Name of application to connect on the RTMP server. This option
613 overrides the parameter specified in the URI.
616 Set the client buffer time in milliseconds. The default is 3000.
619 Extra arbitrary AMF connection parameters, parsed from a string,
620 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
621 Each value is prefixed by a single character denoting the type,
622 B for Boolean, N for number, S for string, O for object, or Z for null,
623 followed by a colon. For Booleans the data must be either 0 or 1 for
624 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
625 1 to end or begin an object, respectively. Data items in subobjects may
626 be named, by prefixing the type with 'N' and specifying the name before
627 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
628 times to construct arbitrary AMF sequences.
631 Version of the Flash plugin used to run the SWF player. The default
632 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
633 <libavformat version>).)
635 @item rtmp_flush_interval
636 Number of packets flushed in the same request (RTMPT only). The default
640 Specify that the media is a live stream. No resuming or seeking in
641 live streams is possible. The default value is @code{any}, which means the
642 subscriber first tries to play the live stream specified in the
643 playpath. If a live stream of that name is not found, it plays the
644 recorded stream. The other possible values are @code{live} and
648 URL of the web page in which the media was embedded. By default no
652 Stream identifier to play or to publish. This option overrides the
653 parameter specified in the URI.
656 Name of live stream to subscribe to. By default no value will be sent.
657 It is only sent if the option is specified or if rtmp_live
661 SHA256 hash of the decompressed SWF file (32 bytes).
664 Size of the decompressed SWF file, required for SWFVerification.
667 URL of the SWF player for the media. By default no value will be sent.
670 URL to player swf file, compute hash/size automatically.
673 URL of the target stream. Defaults to proto://host[:port]/app.
677 For example to read with @command{ffplay} a multimedia resource named
678 "sample" from the application "vod" from an RTMP server "myserver":
680 ffplay rtmp://myserver/vod/sample
683 To publish to a password protected server, passing the playpath and
684 app names separately:
686 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
691 Encrypted Real-Time Messaging Protocol.
693 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
694 streaming multimedia content within standard cryptographic primitives,
695 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
700 Real-Time Messaging Protocol over a secure SSL connection.
702 The Real-Time Messaging Protocol (RTMPS) is used for streaming
703 multimedia content across an encrypted connection.
707 Real-Time Messaging Protocol tunneled through HTTP.
709 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
710 for streaming multimedia content within HTTP requests to traverse
715 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
717 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
718 is used for streaming multimedia content within HTTP requests to traverse
723 Real-Time Messaging Protocol tunneled through HTTPS.
725 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
726 for streaming multimedia content within HTTPS requests to traverse
729 @section libsmbclient
731 libsmbclient permits one to manipulate CIFS/SMB network resources.
733 Following syntax is required.
736 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
739 This protocol accepts the following options.
743 Set timeout in milliseconds of socket I/O operations used by the underlying
744 low level operation. By default it is set to -1, which means that the timeout
748 Truncate existing files on write, if set to 1. A value of 0 prevents
749 truncating. Default value is 1.
752 Set the workgroup used for making connections. By default workgroup is not specified.
756 For more information see: @url{http://www.samba.org/}.
760 Secure File Transfer Protocol via libssh
762 Read from or write to remote resources using SFTP protocol.
764 Following syntax is required.
767 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
770 This protocol accepts the following options.
774 Set timeout of socket I/O operations used by the underlying low level
775 operation. By default it is set to -1, which means that the timeout
779 Truncate existing files on write, if set to 1. A value of 0 prevents
780 truncating. Default value is 1.
783 Specify the path of the file containing private key to use during authorization.
784 By default libssh searches for keys in the @file{~/.ssh/} directory.
788 Example: Play a file stored on remote server.
791 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
794 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
796 Real-Time Messaging Protocol and its variants supported through
799 Requires the presence of the librtmp headers and library during
800 configuration. You need to explicitly configure the build with
801 "--enable-librtmp". If enabled this will replace the native RTMP
804 This protocol provides most client functions and a few server
805 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
806 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
807 variants of these encrypted types (RTMPTE, RTMPTS).
809 The required syntax is:
811 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
814 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
815 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
816 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
817 meaning as specified for the RTMP native protocol.
818 @var{options} contains a list of space-separated options of the form
821 See the librtmp manual page (man 3 librtmp) for more information.
823 For example, to stream a file in real-time to an RTMP server using
826 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
829 To play the same stream using @command{ffplay}:
831 ffplay "rtmp://myserver/live/mystream live=1"
836 Real-time Transport Protocol.
838 The required syntax for an RTP URL is:
839 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
841 @var{port} specifies the RTP port to use.
843 The following URL options are supported:
848 Set the TTL (Time-To-Live) value (for multicast only).
850 @item rtcpport=@var{n}
851 Set the remote RTCP port to @var{n}.
853 @item localrtpport=@var{n}
854 Set the local RTP port to @var{n}.
856 @item localrtcpport=@var{n}'
857 Set the local RTCP port to @var{n}.
859 @item pkt_size=@var{n}
860 Set max packet size (in bytes) to @var{n}.
863 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
866 @item sources=@var{ip}[,@var{ip}]
867 List allowed source IP addresses.
869 @item block=@var{ip}[,@var{ip}]
870 List disallowed (blocked) source IP addresses.
872 @item write_to_source=0|1
873 Send packets to the source address of the latest received packet (if
874 set to 1) or to a default remote address (if set to 0).
876 @item localport=@var{n}
877 Set the local RTP port to @var{n}.
879 This is a deprecated option. Instead, @option{localrtpport} should be
889 If @option{rtcpport} is not set the RTCP port will be set to the RTP
893 If @option{localrtpport} (the local RTP port) is not set any available
894 port will be used for the local RTP and RTCP ports.
897 If @option{localrtcpport} (the local RTCP port) is not set it will be
898 set to the local RTP port value plus 1.
903 Real-Time Streaming Protocol.
905 RTSP is not technically a protocol handler in libavformat, it is a demuxer
906 and muxer. The demuxer supports both normal RTSP (with data transferred
907 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
908 data transferred over RDT).
910 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
911 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
912 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
914 The required syntax for a RTSP url is:
916 rtsp://@var{hostname}[:@var{port}]/@var{path}
919 Options can be set on the @command{ffmpeg}/@command{ffplay} command
920 line, or set in code via @code{AVOption}s or in
921 @code{avformat_open_input}.
923 The following options are supported.
927 Do not start playing the stream immediately if set to 1. Default value
931 Set RTSP transport protocols.
933 It accepts the following values:
936 Use UDP as lower transport protocol.
939 Use TCP (interleaving within the RTSP control channel) as lower
943 Use UDP multicast as lower transport protocol.
946 Use HTTP tunneling as lower transport protocol, which is useful for
950 Multiple lower transport protocols may be specified, in that case they are
951 tried one at a time (if the setup of one fails, the next one is tried).
952 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
957 The following values are accepted:
960 Accept packets only from negotiated peer address and port.
962 Act as a server, listening for an incoming connection.
964 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
967 Default value is @samp{none}.
969 @item allowed_media_types
970 Set media types to accept from the server.
972 The following flags are accepted:
979 By default it accepts all media types.
982 Set minimum local UDP port. Default value is 5000.
985 Set maximum local UDP port. Default value is 65000.
988 Set maximum timeout (in seconds) to wait for incoming connections.
990 A value of -1 means infinite (default). This option implies the
991 @option{rtsp_flags} set to @samp{listen}.
993 @item reorder_queue_size
994 Set number of packets to buffer for handling of reordered packets.
997 Set socket TCP I/O timeout in microseconds.
1000 Override User-Agent header. If not specified, it defaults to the
1001 libavformat identifier string.
1004 When receiving data over UDP, the demuxer tries to reorder received packets
1005 (since they may arrive out of order, or packets may get lost totally). This
1006 can be disabled by setting the maximum demuxing delay to zero (via
1007 the @code{max_delay} field of AVFormatContext).
1009 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1010 streams to display can be chosen with @code{-vst} @var{n} and
1011 @code{-ast} @var{n} for video and audio respectively, and can be switched
1012 on the fly by pressing @code{v} and @code{a}.
1014 @subsection Examples
1016 The following examples all make use of the @command{ffplay} and
1017 @command{ffmpeg} tools.
1021 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1023 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1027 Watch a stream tunneled over HTTP:
1029 ffplay -rtsp_transport http rtsp://server/video.mp4
1033 Send a stream in realtime to a RTSP server, for others to watch:
1035 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1039 Receive a stream in realtime:
1041 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1047 Session Announcement Protocol (RFC 2974). This is not technically a
1048 protocol handler in libavformat, it is a muxer and demuxer.
1049 It is used for signalling of RTP streams, by announcing the SDP for the
1050 streams regularly on a separate port.
1054 The syntax for a SAP url given to the muxer is:
1056 sap://@var{destination}[:@var{port}][?@var{options}]
1059 The RTP packets are sent to @var{destination} on port @var{port},
1060 or to port 5004 if no port is specified.
1061 @var{options} is a @code{&}-separated list. The following options
1066 @item announce_addr=@var{address}
1067 Specify the destination IP address for sending the announcements to.
1068 If omitted, the announcements are sent to the commonly used SAP
1069 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1070 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1072 @item announce_port=@var{port}
1073 Specify the port to send the announcements on, defaults to
1074 9875 if not specified.
1077 Specify the time to live value for the announcements and RTP packets,
1080 @item same_port=@var{0|1}
1081 If set to 1, send all RTP streams on the same port pair. If zero (the
1082 default), all streams are sent on unique ports, with each stream on a
1083 port 2 numbers higher than the previous.
1084 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1085 The RTP stack in libavformat for receiving requires all streams to be sent
1089 Example command lines follow.
1091 To broadcast a stream on the local subnet, for watching in VLC:
1094 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1097 Similarly, for watching in @command{ffplay}:
1100 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1103 And for watching in @command{ffplay}, over IPv6:
1106 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1111 The syntax for a SAP url given to the demuxer is:
1113 sap://[@var{address}][:@var{port}]
1116 @var{address} is the multicast address to listen for announcements on,
1117 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1118 is the port that is listened on, 9875 if omitted.
1120 The demuxers listens for announcements on the given address and port.
1121 Once an announcement is received, it tries to receive that particular stream.
1123 Example command lines follow.
1125 To play back the first stream announced on the normal SAP multicast address:
1131 To play back the first stream announced on one the default IPv6 SAP multicast address:
1134 ffplay sap://[ff0e::2:7ffe]
1139 Stream Control Transmission Protocol.
1141 The accepted URL syntax is:
1143 sctp://@var{host}:@var{port}[?@var{options}]
1146 The protocol accepts the following options:
1149 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1152 Set the maximum number of streams. By default no limit is set.
1157 Secure Real-time Transport Protocol.
1159 The accepted options are:
1162 @item srtp_out_suite
1163 Select input and output encoding suites.
1167 @item AES_CM_128_HMAC_SHA1_80
1168 @item SRTP_AES128_CM_HMAC_SHA1_80
1169 @item AES_CM_128_HMAC_SHA1_32
1170 @item SRTP_AES128_CM_HMAC_SHA1_32
1173 @item srtp_in_params
1174 @item srtp_out_params
1175 Set input and output encoding parameters, which are expressed by a
1176 base64-encoded representation of a binary block. The first 16 bytes of
1177 this binary block are used as master key, the following 14 bytes are
1178 used as master salt.
1183 Virtually extract a segment of a file or another stream.
1184 The underlying stream must be seekable.
1189 Start offset of the extracted segment, in bytes.
1191 End offset of the extracted segment, in bytes.
1192 If set to 0, extract till end of file.
1197 Extract a chapter from a DVD VOB file (start and end sectors obtained
1198 externally and multiplied by 2048):
1200 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1203 Play an AVI file directly from a TAR archive:
1205 subfile,,start,183241728,end,366490624,,:archive.tar
1208 Play a MPEG-TS file from start offset till end:
1210 subfile,,start,32815239,end,0,,:video.ts
1215 Writes the output to multiple protocols. The individual outputs are separated
1219 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1224 Transmission Control Protocol.
1226 The required syntax for a TCP url is:
1228 tcp://@var{hostname}:@var{port}[?@var{options}]
1231 @var{options} contains a list of &-separated options of the form
1232 @var{key}=@var{val}.
1234 The list of supported options follows.
1237 @item listen=@var{1|0}
1238 Listen for an incoming connection. Default value is 0.
1240 @item timeout=@var{microseconds}
1241 Set raise error timeout, expressed in microseconds.
1243 This option is only relevant in read mode: if no data arrived in more
1244 than this time interval, raise error.
1246 @item listen_timeout=@var{milliseconds}
1247 Set listen timeout, expressed in milliseconds.
1249 @item recv_buffer_size=@var{bytes}
1250 Set receive buffer size, expressed bytes.
1252 @item send_buffer_size=@var{bytes}
1253 Set send buffer size, expressed bytes.
1255 @item tcp_nodelay=@var{1|0}
1256 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1259 The following example shows how to setup a listening TCP connection
1260 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1262 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1263 ffplay tcp://@var{hostname}:@var{port}
1268 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1270 The required syntax for a TLS/SSL url is:
1272 tls://@var{hostname}:@var{port}[?@var{options}]
1275 The following parameters can be set via command line options
1276 (or in code via @code{AVOption}s):
1280 @item ca_file, cafile=@var{filename}
1281 A file containing certificate authority (CA) root certificates to treat
1282 as trusted. If the linked TLS library contains a default this might not
1283 need to be specified for verification to work, but not all libraries and
1284 setups have defaults built in.
1285 The file must be in OpenSSL PEM format.
1287 @item tls_verify=@var{1|0}
1288 If enabled, try to verify the peer that we are communicating with.
1289 Note, if using OpenSSL, this currently only makes sure that the
1290 peer certificate is signed by one of the root certificates in the CA
1291 database, but it does not validate that the certificate actually
1292 matches the host name we are trying to connect to. (With other backends,
1293 the host name is validated as well.)
1295 This is disabled by default since it requires a CA database to be
1296 provided by the caller in many cases.
1298 @item cert_file, cert=@var{filename}
1299 A file containing a certificate to use in the handshake with the peer.
1300 (When operating as server, in listen mode, this is more often required
1301 by the peer, while client certificates only are mandated in certain
1304 @item key_file, key=@var{filename}
1305 A file containing the private key for the certificate.
1307 @item listen=@var{1|0}
1308 If enabled, listen for connections on the provided port, and assume
1309 the server role in the handshake instead of the client role.
1313 Example command lines:
1315 To create a TLS/SSL server that serves an input stream.
1318 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1321 To play back a stream from the TLS/SSL server using @command{ffplay}:
1324 ffplay tls://@var{hostname}:@var{port}
1329 User Datagram Protocol.
1331 The required syntax for an UDP URL is:
1333 udp://@var{hostname}:@var{port}[?@var{options}]
1336 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1338 In case threading is enabled on the system, a circular buffer is used
1339 to store the incoming data, which allows one to reduce loss of data due to
1340 UDP socket buffer overruns. The @var{fifo_size} and
1341 @var{overrun_nonfatal} options are related to this buffer.
1343 The list of supported options follows.
1346 @item buffer_size=@var{size}
1347 Set the UDP maximum socket buffer size in bytes. This is used to set either
1348 the receive or send buffer size, depending on what the socket is used for.
1349 Default is 64KB. See also @var{fifo_size}.
1351 @item bitrate=@var{bitrate}
1352 If set to nonzero, the output will have the specified constant bitrate if the
1353 input has enough packets to sustain it.
1355 @item burst_bits=@var{bits}
1356 When using @var{bitrate} this specifies the maximum number of bits in
1359 @item localport=@var{port}
1360 Override the local UDP port to bind with.
1362 @item localaddr=@var{addr}
1363 Choose the local IP address. This is useful e.g. if sending multicast
1364 and the host has multiple interfaces, where the user can choose
1365 which interface to send on by specifying the IP address of that interface.
1367 @item pkt_size=@var{size}
1368 Set the size in bytes of UDP packets.
1370 @item reuse=@var{1|0}
1371 Explicitly allow or disallow reusing UDP sockets.
1374 Set the time to live value (for multicast only).
1376 @item connect=@var{1|0}
1377 Initialize the UDP socket with @code{connect()}. In this case, the
1378 destination address can't be changed with ff_udp_set_remote_url later.
1379 If the destination address isn't known at the start, this option can
1380 be specified in ff_udp_set_remote_url, too.
1381 This allows finding out the source address for the packets with getsockname,
1382 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1383 unreachable" is received.
1384 For receiving, this gives the benefit of only receiving packets from
1385 the specified peer address/port.
1387 @item sources=@var{address}[,@var{address}]
1388 Only receive packets sent to the multicast group from one of the
1389 specified sender IP addresses.
1391 @item block=@var{address}[,@var{address}]
1392 Ignore packets sent to the multicast group from the specified
1393 sender IP addresses.
1395 @item fifo_size=@var{units}
1396 Set the UDP receiving circular buffer size, expressed as a number of
1397 packets with size of 188 bytes. If not specified defaults to 7*4096.
1399 @item overrun_nonfatal=@var{1|0}
1400 Survive in case of UDP receiving circular buffer overrun. Default
1403 @item timeout=@var{microseconds}
1404 Set raise error timeout, expressed in microseconds.
1406 This option is only relevant in read mode: if no data arrived in more
1407 than this time interval, raise error.
1409 @item broadcast=@var{1|0}
1410 Explicitly allow or disallow UDP broadcasting.
1412 Note that broadcasting may not work properly on networks having
1413 a broadcast storm protection.
1416 @subsection Examples
1420 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1422 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1426 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1427 sized UDP packets, using a large input buffer:
1429 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1433 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1435 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1443 The required syntax for a Unix socket URL is:
1446 unix://@var{filepath}
1449 The following parameters can be set via command line options
1450 (or in code via @code{AVOption}s):
1456 Create the Unix socket in listening mode.
1459 @c man end PROTOCOLS