4 Protocols are configured elements in FFmpeg that enable access to
5 resources that require specific protocols.
7 When you configure your FFmpeg build, all the supported protocols are
8 enabled by default. You can list all available ones using the
9 configure option "--list-protocols".
11 You can disable all the protocols using the configure option
12 "--disable-protocols", and selectively enable a protocol using the
13 option "--enable-protocol=@var{PROTOCOL}", or you can disable a
14 particular protocol using the option
15 "--disable-protocol=@var{PROTOCOL}".
17 The option "-protocols" of the ff* tools will display the list of
20 A description of the currently available protocols follows.
26 The accepted options are:
36 Playlist to read (BDMV/PLAYLIST/?????.mpls)
42 Read longest playlist from BluRay mounted to /mnt/bluray:
47 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
49 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
54 Caching wrapper for input stream.
56 Cache the input stream to temporary file. It brings seeking capability to live streams.
64 Physical concatenation protocol.
66 Allow to read and seek from many resource in sequence as if they were
69 A URL accepted by this protocol has the syntax:
71 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
74 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
75 resource to be concatenated, each one possibly specifying a distinct
78 For example to read a sequence of files @file{split1.mpeg},
79 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
82 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
85 Note that you may need to escape the character "|" which is special for
90 AES-encrypted stream reading protocol.
92 The accepted options are:
95 Set the AES decryption key binary block from given hexadecimal representation.
98 Set the AES decryption initialization vector binary block from given hexadecimal representation.
101 Accepted URL formats:
109 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
111 For example, to convert a GIF file given inline with @command{ffmpeg}:
113 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
118 File access protocol.
120 Allow to read from or write to a file.
122 A file URL can have the form:
127 where @var{filename} is the path of the file to read.
129 An URL that does not have a protocol prefix will be assumed to be a
130 file URL. Depending on the build, an URL that looks like a Windows
131 path with the drive letter at the beginning will also be assumed to be
132 a file URL (usually not the case in builds for unix-like systems).
134 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
137 ffmpeg -i file:input.mpeg output.mpeg
140 This protocol accepts the following options:
144 Truncate existing files on write, if set to 1. A value of 0 prevents
145 truncating. Default value is 1.
148 Set I/O operation maximum block size, in bytes. Default value is
149 @code{INT_MAX}, which results in not limiting the requested block size.
150 Setting this value reasonably low improves user termination request reaction
151 time, which is valuable for files on slow medium.
156 FTP (File Transfer Protocol).
158 Allow to read from or write to remote resources using FTP protocol.
160 Following syntax is required.
162 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
165 This protocol accepts the following options.
169 Set timeout in microseconds of socket I/O operations used by the underlying low level
170 operation. By default it is set to -1, which means that the timeout is
173 @item ftp-anonymous-password
174 Password used when login as anonymous user. Typically an e-mail address
177 @item ftp-write-seekable
178 Control seekability of connection during encoding. If set to 1 the
179 resource is supposed to be seekable, if set to 0 it is assumed not
180 to be seekable. Default value is 0.
183 NOTE: Protocol can be used as output, but it is recommended to not do
184 it, unless special care is taken (tests, customized server configuration
185 etc.). Different FTP servers behave in different way during seek
186 operation. ff* tools may produce incomplete content due to server limitations.
194 Read Apple HTTP Live Streaming compliant segmented stream as
195 a uniform one. The M3U8 playlists describing the segments can be
196 remote HTTP resources or local files, accessed using the standard
198 The nested protocol is declared by specifying
199 "+@var{proto}" after the hls URI scheme name, where @var{proto}
200 is either "file" or "http".
203 hls+http://host/path/to/remote/resource.m3u8
204 hls+file://path/to/local/resource.m3u8
207 Using this protocol is discouraged - the hls demuxer should work
208 just as well (if not, please report the issues) and is more complete.
209 To use the hls demuxer instead, simply use the direct URLs to the
214 HTTP (Hyper Text Transfer Protocol).
216 This protocol accepts the following options:
220 Control seekability of connection. If set to 1 the resource is
221 supposed to be seekable, if set to 0 it is assumed not to be seekable,
222 if set to -1 it will try to autodetect if it is seekable. Default
226 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
229 Set a specific content type for the POST messages.
232 Set custom HTTP headers, can override built in default headers. The
233 value must be a string encoding the headers.
235 @item multiple_requests
236 Use persistent connections if set to 1, default is 0.
239 Set custom HTTP post data.
243 Override the User-Agent header. If not specified the protocol will use a
244 string describing the libavformat build. ("Lavf/<version>")
247 Set timeout in microseconds of socket I/O operations used by the underlying low level
248 operation. By default it is set to -1, which means that the timeout is
252 Export the MIME type.
255 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
256 supports this, the metadata has to be retrieved by the application by reading
257 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
260 @item icy_metadata_headers
261 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
262 headers, separated by newline characters.
264 @item icy_metadata_packet
265 If the server supports ICY metadata, and @option{icy} was set to 1, this
266 contains the last non-empty metadata packet sent by the server. It should be
267 polled in regular intervals by applications interested in mid-stream metadata
271 Set the cookies to be sent in future requests. The format of each cookie is the
272 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
273 delimited by a newline character.
276 Set initial byte offset.
279 Try to limit the request to bytes preceding this offset.
282 @subsection HTTP Cookies
284 Some HTTP requests will be denied unless cookie values are passed in with the
285 request. The @option{cookies} option allows these cookies to be specified. At
286 the very least, each cookie must specify a value along with a path and domain.
287 HTTP requests that match both the domain and path will automatically include the
288 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
291 The required syntax to play a stream specifying a cookie is:
293 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
302 Set the genre of the stream.
305 Set the name of the stream.
307 @item ice_description
308 Set the description of the stream.
311 Set the stream website url.
314 Set if the stream should be public.
315 Default is 0 (not public).
318 Password for the mountpoint.
321 If set to 1, enable support for legacy Icecast (Version < 2.4), using the SOURCE method
322 instead of the PUT method.
325 Set a specific content type for the stream.
326 This MUST be set if streaming else than audio/mpeg
329 Override the User-Agent header. If not specified the protocol will use a
330 string describing the libavformat build. ("Lavf/<version>")
335 icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
340 MMS (Microsoft Media Server) protocol over TCP.
344 MMS (Microsoft Media Server) protocol over HTTP.
346 The required syntax is:
348 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
355 Computes the MD5 hash of the data to be written, and on close writes
356 this to the designated output or stdout if none is specified. It can
357 be used to test muxers without writing an actual file.
359 Some examples follow.
361 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
362 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
364 # Write the MD5 hash of the encoded AVI file to stdout.
365 ffmpeg -i input.flv -f avi -y md5:
368 Note that some formats (typically MOV) require the output protocol to
369 be seekable, so they will fail with the MD5 output protocol.
373 UNIX pipe access protocol.
375 Allow to read and write from UNIX pipes.
377 The accepted syntax is:
382 @var{number} is the number corresponding to the file descriptor of the
383 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
384 is not specified, by default the stdout file descriptor will be used
385 for writing, stdin for reading.
387 For example to read from stdin with @command{ffmpeg}:
389 cat test.wav | ffmpeg -i pipe:0
390 # ...this is the same as...
391 cat test.wav | ffmpeg -i pipe:
394 For writing to stdout with @command{ffmpeg}:
396 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
397 # ...this is the same as...
398 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
401 This protocol accepts the following options:
405 Set I/O operation maximum block size, in bytes. Default value is
406 @code{INT_MAX}, which results in not limiting the requested block size.
407 Setting this value reasonably low improves user termination request reaction
408 time, which is valuable if data transmission is slow.
411 Note that some formats (typically MOV), require the output protocol to
412 be seekable, so they will fail with the pipe output protocol.
416 Real-Time Messaging Protocol.
418 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
419 content across a TCP/IP network.
421 The required syntax is:
423 rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
426 The accepted parameters are:
430 An optional username (mostly for publishing).
433 An optional password (mostly for publishing).
436 The address of the RTMP server.
439 The number of the TCP port to use (by default is 1935).
442 It is the name of the application to access. It usually corresponds to
443 the path where the application is installed on the RTMP server
444 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
445 the value parsed from the URI through the @code{rtmp_app} option, too.
448 It is the path or name of the resource to play with reference to the
449 application specified in @var{app}, may be prefixed by "mp4:". You
450 can override the value parsed from the URI through the @code{rtmp_playpath}
454 Act as a server, listening for an incoming connection.
457 Maximum time to wait for the incoming connection. Implies listen.
460 Additionally, the following parameters can be set via command line options
461 (or in code via @code{AVOption}s):
465 Name of application to connect on the RTMP server. This option
466 overrides the parameter specified in the URI.
469 Set the client buffer time in milliseconds. The default is 3000.
472 Extra arbitrary AMF connection parameters, parsed from a string,
473 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
474 Each value is prefixed by a single character denoting the type,
475 B for Boolean, N for number, S for string, O for object, or Z for null,
476 followed by a colon. For Booleans the data must be either 0 or 1 for
477 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
478 1 to end or begin an object, respectively. Data items in subobjects may
479 be named, by prefixing the type with 'N' and specifying the name before
480 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
481 times to construct arbitrary AMF sequences.
484 Version of the Flash plugin used to run the SWF player. The default
485 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
486 <libavformat version>).)
488 @item rtmp_flush_interval
489 Number of packets flushed in the same request (RTMPT only). The default
493 Specify that the media is a live stream. No resuming or seeking in
494 live streams is possible. The default value is @code{any}, which means the
495 subscriber first tries to play the live stream specified in the
496 playpath. If a live stream of that name is not found, it plays the
497 recorded stream. The other possible values are @code{live} and
501 URL of the web page in which the media was embedded. By default no
505 Stream identifier to play or to publish. This option overrides the
506 parameter specified in the URI.
509 Name of live stream to subscribe to. By default no value will be sent.
510 It is only sent if the option is specified or if rtmp_live
514 SHA256 hash of the decompressed SWF file (32 bytes).
517 Size of the decompressed SWF file, required for SWFVerification.
520 URL of the SWF player for the media. By default no value will be sent.
523 URL to player swf file, compute hash/size automatically.
526 URL of the target stream. Defaults to proto://host[:port]/app.
530 For example to read with @command{ffplay} a multimedia resource named
531 "sample" from the application "vod" from an RTMP server "myserver":
533 ffplay rtmp://myserver/vod/sample
536 To publish to a password protected server, passing the playpath and
537 app names separately:
539 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
544 Encrypted Real-Time Messaging Protocol.
546 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
547 streaming multimedia content within standard cryptographic primitives,
548 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
553 Real-Time Messaging Protocol over a secure SSL connection.
555 The Real-Time Messaging Protocol (RTMPS) is used for streaming
556 multimedia content across an encrypted connection.
560 Real-Time Messaging Protocol tunneled through HTTP.
562 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
563 for streaming multimedia content within HTTP requests to traverse
568 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
570 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
571 is used for streaming multimedia content within HTTP requests to traverse
576 Real-Time Messaging Protocol tunneled through HTTPS.
578 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
579 for streaming multimedia content within HTTPS requests to traverse
582 @section libsmbclient
584 libsmbclient permits one to manipulate CIFS/SMB network resources.
586 Following syntax is required.
589 smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
592 This protocol accepts the following options.
596 Set timeout in miliseconds of socket I/O operations used by the underlying
597 low level operation. By default it is set to -1, which means that the timeout
601 Truncate existing files on write, if set to 1. A value of 0 prevents
602 truncating. Default value is 1.
605 Set the workgroup used for making connections. By default workgroup is not specified.
609 For more information see: @url{http://www.samba.org/}.
613 Secure File Transfer Protocol via libssh
615 Allow to read from or write to remote resources using SFTP protocol.
617 Following syntax is required.
620 sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
623 This protocol accepts the following options.
627 Set timeout of socket I/O operations used by the underlying low level
628 operation. By default it is set to -1, which means that the timeout
632 Truncate existing files on write, if set to 1. A value of 0 prevents
633 truncating. Default value is 1.
636 Specify the path of the file containing private key to use during authorization.
637 By default libssh searches for keys in the @file{~/.ssh/} directory.
641 Example: Play a file stored on remote server.
644 ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
647 @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
649 Real-Time Messaging Protocol and its variants supported through
652 Requires the presence of the librtmp headers and library during
653 configuration. You need to explicitly configure the build with
654 "--enable-librtmp". If enabled this will replace the native RTMP
657 This protocol provides most client functions and a few server
658 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
659 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
660 variants of these encrypted types (RTMPTE, RTMPTS).
662 The required syntax is:
664 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
667 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
668 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
669 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
670 meaning as specified for the RTMP native protocol.
671 @var{options} contains a list of space-separated options of the form
674 See the librtmp manual page (man 3 librtmp) for more information.
676 For example, to stream a file in real-time to an RTMP server using
679 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
682 To play the same stream using @command{ffplay}:
684 ffplay "rtmp://myserver/live/mystream live=1"
689 Real-time Transport Protocol.
691 The required syntax for an RTP URL is:
692 rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
694 @var{port} specifies the RTP port to use.
696 The following URL options are supported:
701 Set the TTL (Time-To-Live) value (for multicast only).
703 @item rtcpport=@var{n}
704 Set the remote RTCP port to @var{n}.
706 @item localrtpport=@var{n}
707 Set the local RTP port to @var{n}.
709 @item localrtcpport=@var{n}'
710 Set the local RTCP port to @var{n}.
712 @item pkt_size=@var{n}
713 Set max packet size (in bytes) to @var{n}.
716 Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
719 @item sources=@var{ip}[,@var{ip}]
720 List allowed source IP addresses.
722 @item block=@var{ip}[,@var{ip}]
723 List disallowed (blocked) source IP addresses.
725 @item write_to_source=0|1
726 Send packets to the source address of the latest received packet (if
727 set to 1) or to a default remote address (if set to 0).
729 @item localport=@var{n}
730 Set the local RTP port to @var{n}.
732 This is a deprecated option. Instead, @option{localrtpport} should be
742 If @option{rtcpport} is not set the RTCP port will be set to the RTP
746 If @option{localrtpport} (the local RTP port) is not set any available
747 port will be used for the local RTP and RTCP ports.
750 If @option{localrtcpport} (the local RTCP port) is not set it will be
751 set to the the local RTP port value plus 1.
756 Real-Time Streaming Protocol.
758 RTSP is not technically a protocol handler in libavformat, it is a demuxer
759 and muxer. The demuxer supports both normal RTSP (with data transferred
760 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
761 data transferred over RDT).
763 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
764 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
765 @uref{https://github.com/revmischa/rtsp-server, RTSP server}).
767 The required syntax for a RTSP url is:
769 rtsp://@var{hostname}[:@var{port}]/@var{path}
772 Options can be set on the @command{ffmpeg}/@command{ffplay} command
773 line, or set in code via @code{AVOption}s or in
774 @code{avformat_open_input}.
776 The following options are supported.
780 Do not start playing the stream immediately if set to 1. Default value
784 Set RTSP transport protocols.
786 It accepts the following values:
789 Use UDP as lower transport protocol.
792 Use TCP (interleaving within the RTSP control channel) as lower
796 Use UDP multicast as lower transport protocol.
799 Use HTTP tunneling as lower transport protocol, which is useful for
803 Multiple lower transport protocols may be specified, in that case they are
804 tried one at a time (if the setup of one fails, the next one is tried).
805 For the muxer, only the @samp{tcp} and @samp{udp} options are supported.
810 The following values are accepted:
813 Accept packets only from negotiated peer address and port.
815 Act as a server, listening for an incoming connection.
817 Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
820 Default value is @samp{none}.
822 @item allowed_media_types
823 Set media types to accept from the server.
825 The following flags are accepted:
832 By default it accepts all media types.
835 Set minimum local UDP port. Default value is 5000.
838 Set maximum local UDP port. Default value is 65000.
841 Set maximum timeout (in seconds) to wait for incoming connections.
843 A value of -1 means infinite (default). This option implies the
844 @option{rtsp_flags} set to @samp{listen}.
846 @item reorder_queue_size
847 Set number of packets to buffer for handling of reordered packets.
850 Set socket TCP I/O timeout in microseconds.
853 Override User-Agent header. If not specified, it defaults to the
854 libavformat identifier string.
857 When receiving data over UDP, the demuxer tries to reorder received packets
858 (since they may arrive out of order, or packets may get lost totally). This
859 can be disabled by setting the maximum demuxing delay to zero (via
860 the @code{max_delay} field of AVFormatContext).
862 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
863 streams to display can be chosen with @code{-vst} @var{n} and
864 @code{-ast} @var{n} for video and audio respectively, and can be switched
865 on the fly by pressing @code{v} and @code{a}.
869 The following examples all make use of the @command{ffplay} and
870 @command{ffmpeg} tools.
874 Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
876 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
880 Watch a stream tunneled over HTTP:
882 ffplay -rtsp_transport http rtsp://server/video.mp4
886 Send a stream in realtime to a RTSP server, for others to watch:
888 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
892 Receive a stream in realtime:
894 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
900 Session Announcement Protocol (RFC 2974). This is not technically a
901 protocol handler in libavformat, it is a muxer and demuxer.
902 It is used for signalling of RTP streams, by announcing the SDP for the
903 streams regularly on a separate port.
907 The syntax for a SAP url given to the muxer is:
909 sap://@var{destination}[:@var{port}][?@var{options}]
912 The RTP packets are sent to @var{destination} on port @var{port},
913 or to port 5004 if no port is specified.
914 @var{options} is a @code{&}-separated list. The following options
919 @item announce_addr=@var{address}
920 Specify the destination IP address for sending the announcements to.
921 If omitted, the announcements are sent to the commonly used SAP
922 announcement multicast address 224.2.127.254 (sap.mcast.net), or
923 ff0e::2:7ffe if @var{destination} is an IPv6 address.
925 @item announce_port=@var{port}
926 Specify the port to send the announcements on, defaults to
927 9875 if not specified.
930 Specify the time to live value for the announcements and RTP packets,
933 @item same_port=@var{0|1}
934 If set to 1, send all RTP streams on the same port pair. If zero (the
935 default), all streams are sent on unique ports, with each stream on a
936 port 2 numbers higher than the previous.
937 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
938 The RTP stack in libavformat for receiving requires all streams to be sent
942 Example command lines follow.
944 To broadcast a stream on the local subnet, for watching in VLC:
947 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
950 Similarly, for watching in @command{ffplay}:
953 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
956 And for watching in @command{ffplay}, over IPv6:
959 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
964 The syntax for a SAP url given to the demuxer is:
966 sap://[@var{address}][:@var{port}]
969 @var{address} is the multicast address to listen for announcements on,
970 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
971 is the port that is listened on, 9875 if omitted.
973 The demuxers listens for announcements on the given address and port.
974 Once an announcement is received, it tries to receive that particular stream.
976 Example command lines follow.
978 To play back the first stream announced on the normal SAP multicast address:
984 To play back the first stream announced on one the default IPv6 SAP multicast address:
987 ffplay sap://[ff0e::2:7ffe]
992 Stream Control Transmission Protocol.
994 The accepted URL syntax is:
996 sctp://@var{host}:@var{port}[?@var{options}]
999 The protocol accepts the following options:
1002 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
1005 Set the maximum number of streams. By default no limit is set.
1010 Secure Real-time Transport Protocol.
1012 The accepted options are:
1015 @item srtp_out_suite
1016 Select input and output encoding suites.
1020 @item AES_CM_128_HMAC_SHA1_80
1021 @item SRTP_AES128_CM_HMAC_SHA1_80
1022 @item AES_CM_128_HMAC_SHA1_32
1023 @item SRTP_AES128_CM_HMAC_SHA1_32
1026 @item srtp_in_params
1027 @item srtp_out_params
1028 Set input and output encoding parameters, which are expressed by a
1029 base64-encoded representation of a binary block. The first 16 bytes of
1030 this binary block are used as master key, the following 14 bytes are
1031 used as master salt.
1036 Virtually extract a segment of a file or another stream.
1037 The underlying stream must be seekable.
1042 Start offset of the extracted segment, in bytes.
1044 End offset of the extracted segment, in bytes.
1049 Extract a chapter from a DVD VOB file (start and end sectors obtained
1050 externally and multiplied by 2048):
1052 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1055 Play an AVI file directly from a TAR archive:
1056 subfile,,start,183241728,end,366490624,,:archive.tar
1060 Transmission Control Protocol.
1062 The required syntax for a TCP url is:
1064 tcp://@var{hostname}:@var{port}[?@var{options}]
1067 @var{options} contains a list of &-separated options of the form
1068 @var{key}=@var{val}.
1070 The list of supported options follows.
1073 @item listen=@var{1|0}
1074 Listen for an incoming connection. Default value is 0.
1076 @item timeout=@var{microseconds}
1077 Set raise error timeout, expressed in microseconds.
1079 This option is only relevant in read mode: if no data arrived in more
1080 than this time interval, raise error.
1082 @item listen_timeout=@var{microseconds}
1083 Set listen timeout, expressed in microseconds.
1086 The following example shows how to setup a listening TCP connection
1087 with @command{ffmpeg}, which is then accessed with @command{ffplay}:
1089 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
1090 ffplay tcp://@var{hostname}:@var{port}
1095 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1097 The required syntax for a TLS/SSL url is:
1099 tls://@var{hostname}:@var{port}[?@var{options}]
1102 The following parameters can be set via command line options
1103 (or in code via @code{AVOption}s):
1107 @item ca_file, cafile=@var{filename}
1108 A file containing certificate authority (CA) root certificates to treat
1109 as trusted. If the linked TLS library contains a default this might not
1110 need to be specified for verification to work, but not all libraries and
1111 setups have defaults built in.
1112 The file must be in OpenSSL PEM format.
1114 @item tls_verify=@var{1|0}
1115 If enabled, try to verify the peer that we are communicating with.
1116 Note, if using OpenSSL, this currently only makes sure that the
1117 peer certificate is signed by one of the root certificates in the CA
1118 database, but it does not validate that the certificate actually
1119 matches the host name we are trying to connect to. (With GnuTLS,
1120 the host name is validated as well.)
1122 This is disabled by default since it requires a CA database to be
1123 provided by the caller in many cases.
1125 @item cert_file, cert=@var{filename}
1126 A file containing a certificate to use in the handshake with the peer.
1127 (When operating as server, in listen mode, this is more often required
1128 by the peer, while client certificates only are mandated in certain
1131 @item key_file, key=@var{filename}
1132 A file containing the private key for the certificate.
1134 @item listen=@var{1|0}
1135 If enabled, listen for connections on the provided port, and assume
1136 the server role in the handshake instead of the client role.
1140 Example command lines:
1142 To create a TLS/SSL server that serves an input stream.
1145 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
1148 To play back a stream from the TLS/SSL server using @command{ffplay}:
1151 ffplay tls://@var{hostname}:@var{port}
1156 User Datagram Protocol.
1158 The required syntax for an UDP URL is:
1160 udp://@var{hostname}:@var{port}[?@var{options}]
1163 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
1165 In case threading is enabled on the system, a circular buffer is used
1166 to store the incoming data, which allows one to reduce loss of data due to
1167 UDP socket buffer overruns. The @var{fifo_size} and
1168 @var{overrun_nonfatal} options are related to this buffer.
1170 The list of supported options follows.
1173 @item buffer_size=@var{size}
1174 Set the UDP maximum socket buffer size in bytes. This is used to set either
1175 the receive or send buffer size, depending on what the socket is used for.
1176 Default is 64KB. See also @var{fifo_size}.
1178 @item localport=@var{port}
1179 Override the local UDP port to bind with.
1181 @item localaddr=@var{addr}
1182 Choose the local IP address. This is useful e.g. if sending multicast
1183 and the host has multiple interfaces, where the user can choose
1184 which interface to send on by specifying the IP address of that interface.
1186 @item pkt_size=@var{size}
1187 Set the size in bytes of UDP packets.
1189 @item reuse=@var{1|0}
1190 Explicitly allow or disallow reusing UDP sockets.
1193 Set the time to live value (for multicast only).
1195 @item connect=@var{1|0}
1196 Initialize the UDP socket with @code{connect()}. In this case, the
1197 destination address can't be changed with ff_udp_set_remote_url later.
1198 If the destination address isn't known at the start, this option can
1199 be specified in ff_udp_set_remote_url, too.
1200 This allows finding out the source address for the packets with getsockname,
1201 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1202 unreachable" is received.
1203 For receiving, this gives the benefit of only receiving packets from
1204 the specified peer address/port.
1206 @item sources=@var{address}[,@var{address}]
1207 Only receive packets sent to the multicast group from one of the
1208 specified sender IP addresses.
1210 @item block=@var{address}[,@var{address}]
1211 Ignore packets sent to the multicast group from the specified
1212 sender IP addresses.
1214 @item fifo_size=@var{units}
1215 Set the UDP receiving circular buffer size, expressed as a number of
1216 packets with size of 188 bytes. If not specified defaults to 7*4096.
1218 @item overrun_nonfatal=@var{1|0}
1219 Survive in case of UDP receiving circular buffer overrun. Default
1222 @item timeout=@var{microseconds}
1223 Set raise error timeout, expressed in microseconds.
1225 This option is only relevant in read mode: if no data arrived in more
1226 than this time interval, raise error.
1228 @item broadcast=@var{1|0}
1229 Explicitly allow or disallow UDP broadcasting.
1231 Note that broadcasting may not work properly on networks having
1232 a broadcast storm protection.
1235 @subsection Examples
1239 Use @command{ffmpeg} to stream over UDP to a remote endpoint:
1241 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
1245 Use @command{ffmpeg} to stream in mpegts format over UDP using 188
1246 sized UDP packets, using a large input buffer:
1248 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
1252 Use @command{ffmpeg} to receive over UDP from a remote endpoint:
1254 ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
1262 The required syntax for a Unix socket URL is:
1265 unix://@var{filepath}
1268 The following parameters can be set via command line options
1269 (or in code via @code{AVOption}s):
1275 Create the Unix socket in listening mode.
1278 @c man end PROTOCOLS