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 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
326 supports this, the metadata has to be retrieved by the application by reading
327 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
330 @item icy_metadata_headers
331 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
332 headers, separated by newline characters.
334 @item icy_metadata_packet
335 If the server supports ICY metadata, and @option{icy} was set to 1, this
336 contains the last non-empty metadata packet sent by the server. It should be
337 polled in regular intervals by applications interested in mid-stream metadata
341 Set the cookies to be sent in future requests. The format of each cookie is the
342 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
343 delimited by a newline character.
346 Set initial byte offset.
349 Try to limit the request to bytes preceding this offset.
352 When used as a client option it sets the HTTP method for the request.
354 When used as a server option it sets the HTTP method that is going to be
355 expected from the client(s).
356 If the expected and the received HTTP method do not match the client will
357 be given a Bad Request response.
358 When unset the HTTP method is not checked for now. This will be replaced by
359 autodetection in the future.
362 If set to 1 enables experimental HTTP server. This can be used to send data when
363 used as an output option, or read data from a client with HTTP POST when used as
365 If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
366 in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
368 # Server side (sending):
369 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
371 # Client side (receiving):
372 ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
374 # Client can also be done with wget:
375 wget http://@var{server}:@var{port} -O somefile.ogg
377 # Server side (receiving):
378 ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
380 # Client side (sending):
381 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
383 # Client can also be done with wget:
384 wget --post-file=somefile.ogg http://@var{server}:@var{port}
389 @subsection HTTP Cookies
391 Some HTTP requests will be denied unless cookie values are passed in with the
392 request. The @option{cookies} option allows these cookies to be specified. At
393 the very least, each cookie must specify a value along with a path and domain.
394 HTTP requests that match both the domain and path will automatically include the
395 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
398 The required syntax to play a stream specifying a cookie is:
400 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
405 Icecast protocol (stream to Icecast servers)
407 This protocol accepts the following options:
411 Set the stream genre.
416 @item ice_description
417 Set the stream description.
420 Set the stream website URL.
423 Set if the stream should be public.
424 The default is 0 (not public).
427 Override the User-Agent header. If not specified a string of the form
428 "Lavf/<version>" will be used.
431 Set the Icecast mountpoint password.
434 Set the stream content type. This must be set if it is different from
438 This enables support for Icecast versions < 2.4.0, that do not support the
439 HTTP PUT method but the SOURCE method.
444 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
449 MMS (Microsoft Media Server) protocol over TCP.
453 MMS (Microsoft Media Server) protocol over HTTP.
455 The required syntax is:
457 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
464 Computes the MD5 hash of the data to be written, and on close writes
465 this to the designated output or stdout if none is specified. It can
466 be used to test muxers without writing an actual file.
468 Some examples follow.
470 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
471 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
473 # Write the MD5 hash of the encoded AVI file to stdout.
474 ffmpeg -i input.flv -f avi -y md5:
477 Note that some formats (typically MOV) require the output protocol to
478 be seekable, so they will fail with the MD5 output protocol.
482 UNIX pipe access protocol.
484 Read and write from UNIX pipes.
486 The accepted syntax is:
491 @var{number} is the number corresponding to the file descriptor of the
492 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
493 is not specified, by default the stdout file descriptor will be used
494 for writing, stdin for reading.
496 For example to read from stdin with @command{ffmpeg}:
498 cat test.wav | ffmpeg -i pipe:0
499 # ...this is the same as...
500 cat test.wav | ffmpeg -i pipe:
503 For writing to stdout with @command{ffmpeg}:
505 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
506 # ...this is the same as...
507 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
510 This protocol accepts the following options:
514 Set I/O operation maximum block size, in bytes. Default value is
515 @code{INT_MAX}, which results in not limiting the requested block size.
516 Setting this value reasonably low improves user termination request reaction
517 time, which is valuable if data transmission is slow.
520 Note that some formats (typically MOV), require the output protocol to
521 be seekable, so they will fail with the pipe output protocol.
525 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
527 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
528 for MPEG-2 Transport Streams sent over RTP.
530 This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
531 the @code{rtp} protocol.
533 The required syntax is:
535 -f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
538 The destination UDP ports are @code{port + 2} for the column FEC stream
539 and @code{port + 4} for the row FEC stream.
541 This protocol accepts the following options:
545 The number of columns (4-20, LxD <= 100)
548 The number of rows (4-20, LxD <= 100)
555 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
560 Real-Time Messaging Protocol.
562 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
563 content across a TCP/IP network.
565 The required syntax is:
567 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
570 The accepted parameters are:
574 An optional username (mostly for publishing).
577 An optional password (mostly for publishing).
580 The address of the RTMP server.
583 The number of the TCP port to use (by default is 1935).
586 It is the name of the application to access. It usually corresponds to
587 the path where the application is installed on the RTMP server
588 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
589 the value parsed from the URI through the @code{rtmp_app} option, too.
592 It is the path or name of the resource to play with reference to the
593 application specified in @var{app}, may be prefixed by "mp4:". You
594 can override the value parsed from the URI through the @code{rtmp_playpath}
598 Act as a server, listening for an incoming connection.
601 Maximum time to wait for the incoming connection. Implies listen.
604 Additionally, the following parameters can be set via command line options
605 (or in code via @code{AVOption}s):
609 Name of application to connect on the RTMP server. This option
610 overrides the parameter specified in the URI.
613 Set the client buffer time in milliseconds. The default is 3000.
616 Extra arbitrary AMF connection parameters, parsed from a string,
617 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
618 Each value is prefixed by a single character denoting the type,
619 B for Boolean, N for number, S for string, O for object, or Z for null,
620 followed by a colon. For Booleans the data must be either 0 or 1 for
621 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
622 1 to end or begin an object, respectively. Data items in subobjects may
623 be named, by prefixing the type with 'N' and specifying the name before
624 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
625 times to construct arbitrary AMF sequences.
628 Version of the Flash plugin used to run the SWF player. The default
629 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
630 <libavformat version>).)
632 @item rtmp_flush_interval
633 Number of packets flushed in the same request (RTMPT only). The default
637 Specify that the media is a live stream. No resuming or seeking in
638 live streams is possible. The default value is @code{any}, which means the
639 subscriber first tries to play the live stream specified in the
640 playpath. If a live stream of that name is not found, it plays the
641 recorded stream. The other possible values are @code{live} and
645 URL of the web page in which the media was embedded. By default no
649 Stream identifier to play or to publish. This option overrides the
650 parameter specified in the URI.
653 Name of live stream to subscribe to. By default no value will be sent.
654 It is only sent if the option is specified or if rtmp_live
658 SHA256 hash of the decompressed SWF file (32 bytes).
661 Size of the decompressed SWF file, required for SWFVerification.
664 URL of the SWF player for the media. By default no value will be sent.
667 URL to player swf file, compute hash/size automatically.
670 URL of the target stream. Defaults to proto://host[:port]/app.
674 For example to read with @command{ffplay} a multimedia resource named
675 "sample" from the application "vod" from an RTMP server "myserver":
677 ffplay rtmp://myserver/vod/sample
680 To publish to a password protected server, passing the playpath and
681 app names separately:
683 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
688 Encrypted Real-Time Messaging Protocol.
690 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
691 streaming multimedia content within standard cryptographic primitives,
692 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
697 Real-Time Messaging Protocol over a secure SSL connection.
699 The Real-Time Messaging Protocol (RTMPS) is used for streaming
700 multimedia content across an encrypted connection.
704 Real-Time Messaging Protocol tunneled through HTTP.
706 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
707 for streaming multimedia content within HTTP requests to traverse
712 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
714 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
715 is used for streaming multimedia content within HTTP requests to traverse
720 Real-Time Messaging Protocol tunneled through HTTPS.
722 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
723 for streaming multimedia content within HTTPS requests to traverse
726 @section libsmbclient
728 libsmbclient permits one to manipulate CIFS/SMB network resources.
730 Following syntax is required.
733 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
736 This protocol accepts the following options.
740 Set timeout in milliseconds of socket I/O operations used by the underlying
741 low level operation. By default it is set to -1, which means that the timeout
745 Truncate existing files on write, if set to 1. A value of 0 prevents
746 truncating. Default value is 1.
749 Set the workgroup used for making connections. By default workgroup is not specified.
753 For more information see: @url{http://www.samba.org/}.
757 Secure File Transfer Protocol via libssh
759 Read from or write to remote resources using SFTP protocol.
761 Following syntax is required.
764 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
767 This protocol accepts the following options.
771 Set timeout of socket I/O operations used by the underlying low level
772 operation. By default it is set to -1, which means that the timeout
776 Truncate existing files on write, if set to 1. A value of 0 prevents
777 truncating. Default value is 1.
780 Specify the path of the file containing private key to use during authorization.
781 By default libssh searches for keys in the @file{~/.ssh/} directory.
785 Example: Play a file stored on remote server.
788 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
791 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
793 Real-Time Messaging Protocol and its variants supported through
796 Requires the presence of the librtmp headers and library during
797 configuration. You need to explicitly configure the build with
798 "--enable-librtmp". If enabled this will replace the native RTMP
801 This protocol provides most client functions and a few server
802 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
803 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
804 variants of these encrypted types (RTMPTE, RTMPTS).
806 The required syntax is:
808 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
811 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
812 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
813 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
814 meaning as specified for the RTMP native protocol.
815 @var{options} contains a list of space-separated options of the form
818 See the librtmp manual page (man 3 librtmp) for more information.
820 For example, to stream a file in real-time to an RTMP server using
823 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
826 To play the same stream using @command{ffplay}:
828 ffplay "rtmp://myserver/live/mystream live=1"
833 Real-time Transport Protocol.
835 The required syntax for an RTP URL is:
836 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
838 @var{port} specifies the RTP port to use.
840 The following URL options are supported:
845 Set the TTL (Time-To-Live) value (for multicast only).
847 @item rtcpport=@var{n}
848 Set the remote RTCP port to @var{n}.
850 @item localrtpport=@var{n}
851 Set the local RTP port to @var{n}.
853 @item localrtcpport=@var{n}'
854 Set the local RTCP port to @var{n}.
856 @item pkt_size=@var{n}
857 Set max packet size (in bytes) to @var{n}.
860 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
863 @item sources=@var{ip}[,@var{ip}]
864 List allowed source IP addresses.
866 @item block=@var{ip}[,@var{ip}]
867 List disallowed (blocked) source IP addresses.
869 @item write_to_source=0|1
870 Send packets to the source address of the latest received packet (if
871 set to 1) or to a default remote address (if set to 0).
873 @item localport=@var{n}
874 Set the local RTP port to @var{n}.
876 This is a deprecated option. Instead, @option{localrtpport} should be
886 If @option{rtcpport} is not set the RTCP port will be set to the RTP
890 If @option{localrtpport} (the local RTP port) is not set any available
891 port will be used for the local RTP and RTCP ports.
894 If @option{localrtcpport} (the local RTCP port) is not set it will be
895 set to the local RTP port value plus 1.
900 Real-Time Streaming Protocol.
902 RTSP is not technically a protocol handler in libavformat, it is a demuxer
903 and muxer. The demuxer supports both normal RTSP (with data transferred
904 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
905 data transferred over RDT).
907 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
908 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
909 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
911 The required syntax for a RTSP url is:
913 rtsp://@var{hostname}[:@var{port}]/@var{path}
916 Options can be set on the @command{ffmpeg}/@command{ffplay} command
917 line, or set in code via @code{AVOption}s or in
918 @code{avformat_open_input}.
920 The following options are supported.
924 Do not start playing the stream immediately if set to 1. Default value
928 Set RTSP transport protocols.
930 It accepts the following values:
933 Use UDP as lower transport protocol.
936 Use TCP (interleaving within the RTSP control channel) as lower
940 Use UDP multicast as lower transport protocol.
943 Use HTTP tunneling as lower transport protocol, which is useful for
947 Multiple lower transport protocols may be specified, in that case they are
948 tried one at a time (if the setup of one fails, the next one is tried).
949 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
954 The following values are accepted:
957 Accept packets only from negotiated peer address and port.
959 Act as a server, listening for an incoming connection.
961 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
964 Default value is @samp{none}.
966 @item allowed_media_types
967 Set media types to accept from the server.
969 The following flags are accepted:
976 By default it accepts all media types.
979 Set minimum local UDP port. Default value is 5000.
982 Set maximum local UDP port. Default value is 65000.
985 Set maximum timeout (in seconds) to wait for incoming connections.
987 A value of -1 means infinite (default). This option implies the
988 @option{rtsp_flags} set to @samp{listen}.
990 @item reorder_queue_size
991 Set number of packets to buffer for handling of reordered packets.
994 Set socket TCP I/O timeout in microseconds.
997 Override User-Agent header. If not specified, it defaults to the
998 libavformat identifier string.
1001 When receiving data over UDP, the demuxer tries to reorder received packets
1002 (since they may arrive out of order, or packets may get lost totally). This
1003 can be disabled by setting the maximum demuxing delay to zero (via
1004 the @code{max_delay} field of AVFormatContext).
1006 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
1007 streams to display can be chosen with @code{-vst} @var{n} and
1008 @code{-ast} @var{n} for video and audio respectively, and can be switched
1009 on the fly by pressing @code{v} and @code{a}.
1011 @subsection Examples
1013 The following examples all make use of the @command{ffplay} and
1014 @command{ffmpeg} tools.
1018 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
1020 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1024 Watch a stream tunneled over HTTP:
1026 ffplay -rtsp_transport http rtsp://server/video.mp4
1030 Send a stream in realtime to a RTSP server, for others to watch:
1032 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1036 Receive a stream in realtime:
1038 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
1044 Session Announcement Protocol (RFC 2974). This is not technically a
1045 protocol handler in libavformat, it is a muxer and demuxer.
1046 It is used for signalling of RTP streams, by announcing the SDP for the
1047 streams regularly on a separate port.
1051 The syntax for a SAP url given to the muxer is:
1053 sap://@var{destination}[:@var{port}][?@var{options}]
1056 The RTP packets are sent to @var{destination} on port @var{port},
1057 or to port 5004 if no port is specified.
1058 @var{options} is a @code{&}-separated list. The following options
1063 @item announce_addr=@var{address}
1064 Specify the destination IP address for sending the announcements to.
1065 If omitted, the announcements are sent to the commonly used SAP
1066 announcement multicast address 224.2.127.254 (sap.mcast.net), or
1067 ff0e::2:7ffe if @var{destination} is an IPv6 address.
1069 @item announce_port=@var{port}
1070 Specify the port to send the announcements on, defaults to
1071 9875 if not specified.
1074 Specify the time to live value for the announcements and RTP packets,
1077 @item same_port=@var{0|1}
1078 If set to 1, send all RTP streams on the same port pair. If zero (the
1079 default), all streams are sent on unique ports, with each stream on a
1080 port 2 numbers higher than the previous.
1081 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
1082 The RTP stack in libavformat for receiving requires all streams to be sent
1086 Example command lines follow.
1088 To broadcast a stream on the local subnet, for watching in VLC:
1091 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
1094 Similarly, for watching in @command{ffplay}:
1097 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
1100 And for watching in @command{ffplay}, over IPv6:
1103 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
1108 The syntax for a SAP url given to the demuxer is:
1110 sap://[@var{address}][:@var{port}]
1113 @var{address} is the multicast address to listen for announcements on,
1114 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
1115 is the port that is listened on, 9875 if omitted.
1117 The demuxers listens for announcements on the given address and port.
1118 Once an announcement is received, it tries to receive that particular stream.
1120 Example command lines follow.
1122 To play back the first stream announced on the normal SAP multicast address:
1128 To play back the first stream announced on one the default IPv6 SAP multicast address:
1131 ffplay sap://[ff0e::2:7ffe]
1136 Stream Control Transmission Protocol.
1138 The accepted URL syntax is:
1140 sctp://@var{host}:@var{port}[?@var{options}]
1143 The protocol accepts the following options:
1146 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1149 Set the maximum number of streams. By default no limit is set.
1154 Secure Real-time Transport Protocol.
1156 The accepted options are:
1159 @item srtp_out_suite
1160 Select input and output encoding suites.
1164 @item AES_CM_128_HMAC_SHA1_80
1165 @item SRTP_AES128_CM_HMAC_SHA1_80
1166 @item AES_CM_128_HMAC_SHA1_32
1167 @item SRTP_AES128_CM_HMAC_SHA1_32
1170 @item srtp_in_params
1171 @item srtp_out_params
1172 Set input and output encoding parameters, which are expressed by a
1173 base64-encoded representation of a binary block. The first 16 bytes of
1174 this binary block are used as master key, the following 14 bytes are
1175 used as master salt.
1180 Virtually extract a segment of a file or another stream.
1181 The underlying stream must be seekable.
1186 Start offset of the extracted segment, in bytes.
1188 End offset of the extracted segment, in bytes.
1189 If set to 0, extract till end of file.
1194 Extract a chapter from a DVD VOB file (start and end sectors obtained
1195 externally and multiplied by 2048):
1197 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1200 Play an AVI file directly from a TAR archive:
1202 subfile,,start,183241728,end,366490624,,:archive.tar
1205 Play a MPEG-TS file from start offset till end:
1207 subfile,,start,32815239,end,0,,:video.ts
1212 Writes the output to multiple protocols. The individual outputs are separated
1216 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1221 Transmission Control Protocol.
1223 The required syntax for a TCP url is:
1225 tcp://@var{hostname}:@var{port}[?@var{options}]
1228 @var{options} contains a list of &-separated options of the form
1229 @var{key}=@var{val}.
1231 The list of supported options follows.
1234 @item listen=@var{1|0}
1235 Listen for an incoming connection. Default value is 0.
1237 @item timeout=@var{microseconds}
1238 Set raise error timeout, expressed in microseconds.
1240 This option is only relevant in read mode: if no data arrived in more
1241 than this time interval, raise error.
1243 @item listen_timeout=@var{milliseconds}
1244 Set listen timeout, expressed in milliseconds.
1246 @item recv_buffer_size=@var{bytes}
1247 Set receive buffer size, expressed bytes.
1249 @item send_buffer_size=@var{bytes}
1250 Set send buffer size, expressed bytes.
1252 @item tcp_nodelay=@var{1|0}
1253 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1256 The following example shows how to setup a listening TCP connection
1257 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1259 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1260 ffplay tcp://@var{hostname}:@var{port}
1265 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1267 The required syntax for a TLS/SSL url is:
1269 tls://@var{hostname}:@var{port}[?@var{options}]
1272 The following parameters can be set via command line options
1273 (or in code via @code{AVOption}s):
1277 @item ca_file, cafile=@var{filename}
1278 A file containing certificate authority (CA) root certificates to treat
1279 as trusted. If the linked TLS library contains a default this might not
1280 need to be specified for verification to work, but not all libraries and
1281 setups have defaults built in.
1282 The file must be in OpenSSL PEM format.
1284 @item tls_verify=@var{1|0}
1285 If enabled, try to verify the peer that we are communicating with.
1286 Note, if using OpenSSL, this currently only makes sure that the
1287 peer certificate is signed by one of the root certificates in the CA
1288 database, but it does not validate that the certificate actually
1289 matches the host name we are trying to connect to. (With GnuTLS,
1290 the host name is validated as well.)
1292 This is disabled by default since it requires a CA database to be
1293 provided by the caller in many cases.
1295 @item cert_file, cert=@var{filename}
1296 A file containing a certificate to use in the handshake with the peer.
1297 (When operating as server, in listen mode, this is more often required
1298 by the peer, while client certificates only are mandated in certain
1301 @item key_file, key=@var{filename}
1302 A file containing the private key for the certificate.
1304 @item listen=@var{1|0}
1305 If enabled, listen for connections on the provided port, and assume
1306 the server role in the handshake instead of the client role.
1310 Example command lines:
1312 To create a TLS/SSL server that serves an input stream.
1315 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1318 To play back a stream from the TLS/SSL server using @command{ffplay}:
1321 ffplay tls://@var{hostname}:@var{port}
1326 User Datagram Protocol.
1328 The required syntax for an UDP URL is:
1330 udp://@var{hostname}:@var{port}[?@var{options}]
1333 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1335 In case threading is enabled on the system, a circular buffer is used
1336 to store the incoming data, which allows one to reduce loss of data due to
1337 UDP socket buffer overruns. The @var{fifo_size} and
1338 @var{overrun_nonfatal} options are related to this buffer.
1340 The list of supported options follows.
1343 @item buffer_size=@var{size}
1344 Set the UDP maximum socket buffer size in bytes. This is used to set either
1345 the receive or send buffer size, depending on what the socket is used for.
1346 Default is 64KB. See also @var{fifo_size}.
1348 @item bitrate=@var{bitrate}
1349 If set to nonzero, the output will have the specified constant bitrate if the
1350 input has enough packets to sustain it.
1352 @item burst_bits=@var{bits}
1353 When using @var{bitrate} this specifies the maximum number of bits in
1356 @item localport=@var{port}
1357 Override the local UDP port to bind with.
1359 @item localaddr=@var{addr}
1360 Choose the local IP address. This is useful e.g. if sending multicast
1361 and the host has multiple interfaces, where the user can choose
1362 which interface to send on by specifying the IP address of that interface.
1364 @item pkt_size=@var{size}
1365 Set the size in bytes of UDP packets.
1367 @item reuse=@var{1|0}
1368 Explicitly allow or disallow reusing UDP sockets.
1371 Set the time to live value (for multicast only).
1373 @item connect=@var{1|0}
1374 Initialize the UDP socket with @code{connect()}. In this case, the
1375 destination address can't be changed with ff_udp_set_remote_url later.
1376 If the destination address isn't known at the start, this option can
1377 be specified in ff_udp_set_remote_url, too.
1378 This allows finding out the source address for the packets with getsockname,
1379 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1380 unreachable" is received.
1381 For receiving, this gives the benefit of only receiving packets from
1382 the specified peer address/port.
1384 @item sources=@var{address}[,@var{address}]
1385 Only receive packets sent to the multicast group from one of the
1386 specified sender IP addresses.
1388 @item block=@var{address}[,@var{address}]
1389 Ignore packets sent to the multicast group from the specified
1390 sender IP addresses.
1392 @item fifo_size=@var{units}
1393 Set the UDP receiving circular buffer size, expressed as a number of
1394 packets with size of 188 bytes. If not specified defaults to 7*4096.
1396 @item overrun_nonfatal=@var{1|0}
1397 Survive in case of UDP receiving circular buffer overrun. Default
1400 @item timeout=@var{microseconds}
1401 Set raise error timeout, expressed in microseconds.
1403 This option is only relevant in read mode: if no data arrived in more
1404 than this time interval, raise error.
1406 @item broadcast=@var{1|0}
1407 Explicitly allow or disallow UDP broadcasting.
1409 Note that broadcasting may not work properly on networks having
1410 a broadcast storm protection.
1413 @subsection Examples
1417 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1419 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1423 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1424 sized UDP packets, using a large input buffer:
1426 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1430 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1432 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1440 The required syntax for a Unix socket URL is:
1443 unix://@var{filepath}
1446 The following parameters can be set via command line options
1447 (or in code via @code{AVOption}s):
1453 Create the Unix socket in listening mode.
1456 @c man end PROTOCOLS