1FFMPEG-PROTOCOLS(1) FFMPEG-PROTOCOLS(1)
2
6 ffmpeg-protocols - FFmpeg protocols
7
9 This document describes the input and output protocols provided by the
10 libavformat library.
11
13 The libavformat library provides some generic global options, which can
14 be set on all the protocols. In addition each protocol may support so-
15 called private options, which are specific for that component.
16 Options may be set by specifying -option value in the FFmpeg tools, or
18 by setting the value explicitly in the "AVFormatContext" options or
19 using the libavutil/opt.h API for programmatic use.
20 The list of supported options follows:
22 protocol_whitelist list (input)
24 Set a ","-separated list of allowed protocols. "ALL" matches all
25 protocols. Protocols prefixed by "-" are disabled. All protocols
26 are allowed by default but protocols used by an another protocol
27 (nested protocols) are restricted to a per protocol subset.
28
30 Protocols are configured elements in FFmpeg that enable access to
31 resources that require specific protocols.
32 When you configure your FFmpeg build, all the supported protocols are
34 enabled by default. You can list all available ones using the configure
35 option "--list-protocols".
36 You can disable all the protocols using the configure option
38 "--disable-protocols", and selectively enable a protocol using the
39 option "--enable-protocol=PROTOCOL", or you can disable a particular
40 protocol using the option "--disable-protocol=PROTOCOL".
41 The option "-protocols" of the ff* tools will display the list of
43 supported protocols.
44 All protocols accept the following options:
46 rw_timeout
48 Maximum time to wait for (network) read/write operations to
49 complete, in microseconds.
50 A description of the currently available protocols follows.
52 amqp
54 Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker
55 based publish-subscribe communication protocol.
56 FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A
58 separate AMQP broker must also be run. An example open-source AMQP
59 broker is RabbitMQ.
60 After starting the broker, an FFmpeg client may stream data to the
62 broker using the command:
63 ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost]
65 Where hostname and port (default is 5672) is the address of the broker.
67 The client may also set a user/password for authentication. The default
68 for both fields is "guest". Name of virtual host on broker can be set
69 with vhost. The default value is "/".
70 Muliple subscribers may stream from the broker using the command:
72 ffplay amqp://[[user]:[password]@]hostname[:port][/vhost]
74 In RabbitMQ all data published to the broker flows through a specific
76 exchange, and each subscribing client has an assigned queue/buffer.
77 When a packet arrives at an exchange, it may be copied to a client's
78 queue depending on the exchange and routing_key fields.
79 The following options are supported:
81 exchange
83 Sets the exchange to use on the broker. RabbitMQ has several
84 predefined exchanges: "amq.direct" is the default exchange, where
85 the publisher and subscriber must have a matching routing_key;
86 "amq.fanout" is the same as a broadcast operation (i.e. the data is
87 forwarded to all queues on the fanout exchange independent of the
88 routing_key); and "amq.topic" is similar to "amq.direct", but
89 allows for more complex pattern matching (refer to the RabbitMQ
90 documentation).
91 routing_key
93 Sets the routing key. The default value is "amqp". The routing key
94 is used on the "amq.direct" and "amq.topic" exchanges to decide
95 whether packets are written to the queue of a subscriber.
96 pkt_size
98 Maximum size of each packet sent/received to the broker. Default is
99 131072. Minimum is 4096 and max is any large value (representable
100 by an int). When receiving packets, this sets an internal buffer
101 size in FFmpeg. It should be equal to or greater than the size of
102 the published packets to the broker. Otherwise the received message
103 may be truncated causing decoding errors.
104 connection_timeout
106 The timeout in seconds during the initial connection to the broker.
107 The default value is rw_timeout, or 5 seconds if rw_timeout is not
108 set.
109 delivery_mode mode
111 Sets the delivery mode of each message sent to broker. The
112 following values are accepted:
113 persistent
115 Delivery mode set to "persistent" (2). This is the default
116 value. Messages may be written to the broker's disk depending
117 on its setup.
118 non-persistent
120 Delivery mode set to "non-persistent" (1). Messages will stay
121 in broker's memory unless the broker is under memory pressure.
122 async
124 Asynchronous data filling wrapper for input stream.
125 Fill data in a background thread, to decouple I/O operation from demux
127 thread.
128 async:<URL>
130 async:http://host/resource
131 async:cache:http://host/resource
132 bluray
134 Read BluRay playlist.
135 The accepted options are:
137 angle
139 BluRay angle
140 chapter
142 Start chapter (1...N)
143 playlist
145 Playlist to read (BDMV/PLAYLIST/?????.mpls)
146 Examples:
148 Read longest playlist from BluRay mounted to /mnt/bluray:
150 bluray:/mnt/bluray
152 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start
154 from chapter 2:
155 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
157 cache
159 Caching wrapper for input stream.
160 Cache the input stream to temporary file. It brings seeking capability
162 to live streams.
163 The accepted options are:
165 read_ahead_limit
167 Amount in bytes that may be read ahead when seeking isn't
168 supported. Range is -1 to INT_MAX. -1 for unlimited. Default is
169 65536.
170 URL Syntax is
172 cache:<URL>
174 concat
176 Physical concatenation protocol.
177 Read and seek from many resources in sequence as if they were a unique
179 resource.
180 A URL accepted by this protocol has the syntax:
182 concat:<URL1>|<URL2>|...|<URLN>
184 where URL1, URL2, ..., URLN are the urls of the resource to be
186 concatenated, each one possibly specifying a distinct protocol.
187 For example to read a sequence of files split1.mpeg, split2.mpeg,
189 split3.mpeg with ffplay use the command:
190 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
192 Note that you may need to escape the character "|" which is special for
194 many shells.
195 concatf
197 Physical concatenation protocol using a line break delimited list of
198 resources.
199 Read and seek from many resources in sequence as if they were a unique
201 resource.
202 A URL accepted by this protocol has the syntax:
204 concatf:<URL>
206 where URL is the url containing a line break delimited list of
208 resources to be concatenated, each one possibly specifying a distinct
209 protocol. Special characters must be escaped with backslash or single
210 quotes. See the "Quoting and escaping" section in the ffmpeg-utils(1)
211 manual.
212 For example to read a sequence of files split1.mpeg, split2.mpeg,
214 split3.mpeg listed in separate lines within a file split.txt with
215 ffplay use the command:
216 ffplay concatf:split.txt
218 Where split.txt contains the lines:
220 split1.mpeg
222 split2.mpeg
223 split3.mpeg
224 crypto
226 AES-encrypted stream reading protocol.
227 The accepted options are:
229 key Set the AES decryption key binary block from given hexadecimal
231 representation.
232 iv Set the AES decryption initialization vector binary block from
234 given hexadecimal representation.
235 Accepted URL formats:
237 crypto:<URL>
239 crypto+<URL>
240 data
242 Data in-line in the URI. See
243 <http://en.wikipedia.org/wiki/Data_URI_scheme>.
244 For example, to convert a GIF file given inline with ffmpeg:
246 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
248 fd
250 File descriptor access protocol.
251 The accepted syntax is:
253 fd: -fd <file_descriptor>
255 If fd is not specified, by default the stdout file descriptor will be
257 used for writing, stdin for reading. Unlike the pipe protocol, fd
258 protocol has seek support if it corresponding to a regular file. fd
259 protocol doesn't support pass file descriptor via URL for security.
260 This protocol accepts the following options:
262 blocksize
264 Set I/O operation maximum block size, in bytes. Default value is
265 "INT_MAX", which results in not limiting the requested block size.
266 Setting this value reasonably low improves user termination request
267 reaction time, which is valuable if data transmission is slow.
268 fd Set file descriptor.
270 file
272 File access protocol.
273 Read from or write to a file.
275 A file URL can have the form:
277 file:<filename>
279 where filename is the path of the file to read.
281 An URL that does not have a protocol prefix will be assumed to be a
283 file URL. Depending on the build, an URL that looks like a Windows path
284 with the drive letter at the beginning will also be assumed to be a
285 file URL (usually not the case in builds for unix-like systems).
286 For example to read from a file input.mpeg with ffmpeg use the command:
288 ffmpeg -i file:input.mpeg output.mpeg
290 This protocol accepts the following options:
292 truncate
294 Truncate existing files on write, if set to 1. A value of 0
295 prevents truncating. Default value is 1.
296 blocksize
298 Set I/O operation maximum block size, in bytes. Default value is
299 "INT_MAX", which results in not limiting the requested block size.
300 Setting this value reasonably low improves user termination request
301 reaction time, which is valuable for files on slow medium.
302 follow
304 If set to 1, the protocol will retry reading at the end of the
305 file, allowing reading files that still are being written. In order
306 for this to terminate, you either need to use the rw_timeout
307 option, or use the interrupt callback (for API users).
308 seekable
310 Controls if seekability is advertised on the file. 0 means non-
311 seekable, -1 means auto (seekable for normal files, non-seekable
312 for named pipes).
313 Many demuxers handle seekable and non-seekable resources
315 differently, overriding this might speed up opening certain files
316 at the cost of losing some features (e.g. accurate seeking).
317 ftp
319 FTP (File Transfer Protocol).
320 Read from or write to remote resources using FTP protocol.
322 Following syntax is required.
324 ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
326 This protocol accepts the following options.
328 timeout
330 Set timeout in microseconds of socket I/O operations used by the
331 underlying low level operation. By default it is set to -1, which
332 means that the timeout is not specified.
333 ftp-user
335 Set a user to be used for authenticating to the FTP server. This is
336 overridden by the user in the FTP URL.
337 ftp-password
339 Set a password to be used for authenticating to the FTP server.
340 This is overridden by the password in the FTP URL, or by ftp-
341 anonymous-password if no user is set.
342 ftp-anonymous-password
344 Password used when login as anonymous user. Typically an e-mail
345 address should be used.
346 ftp-write-seekable
348 Control seekability of connection during encoding. If set to 1 the
349 resource is supposed to be seekable, if set to 0 it is assumed not
350 to be seekable. Default value is 0.
351 NOTE: Protocol can be used as output, but it is recommended to not do
353 it, unless special care is taken (tests, customized server
354 configuration etc.). Different FTP servers behave in different way
355 during seek operation. ff* tools may produce incomplete content due to
356 server limitations.
357 gopher
359 Gopher protocol.
360 gophers
362 Gophers protocol.
363 The Gopher protocol with TLS encapsulation.
365 hls
367 Read Apple HTTP Live Streaming compliant segmented stream as a uniform
368 one. The M3U8 playlists describing the segments can be remote HTTP
369 resources or local files, accessed using the standard file protocol.
370 The nested protocol is declared by specifying "+proto" after the hls
371 URI scheme name, where proto is either "file" or "http".
372 hls+http://host/path/to/remote/resource.m3u8
374 hls+file://path/to/local/resource.m3u8
375 Using this protocol is discouraged - the hls demuxer should work just
377 as well (if not, please report the issues) and is more complete. To
378 use the hls demuxer instead, simply use the direct URLs to the m3u8
379 files.
380 http
382 HTTP (Hyper Text Transfer Protocol).
383 This protocol accepts the following options:
385 seekable
387 Control seekability of connection. If set to 1 the resource is
388 supposed to be seekable, if set to 0 it is assumed not to be
389 seekable, if set to -1 it will try to autodetect if it is seekable.
390 Default value is -1.
391 chunked_post
393 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
394 content_type
396 Set a specific content type for the POST messages or for listen
397 mode.
398 http_proxy
400 set HTTP proxy to tunnel through e.g. http://example.com:1234
401 headers
403 Set custom HTTP headers, can override built in default headers. The
404 value must be a string encoding the headers.
405 multiple_requests
407 Use persistent connections if set to 1, default is 0.
408 post_data
410 Set custom HTTP post data.
411 referer
413 Set the Referer header. Include 'Referer: URL' header in HTTP
414 request.
415 user_agent
417 Override the User-Agent header. If not specified the protocol will
418 use a string describing the libavformat build. ("Lavf/<version>")
419 reconnect_at_eof
421 If set then eof is treated like an error and causes reconnection,
422 this is useful for live / endless streams.
423 reconnect_streamed
425 If set then even streamed/non seekable streams will be reconnected
426 on errors.
427 reconnect_on_network_error
429 Reconnect automatically in case of TCP/TLS errors during connect.
430 reconnect_on_http_error
432 A comma separated list of HTTP status codes to reconnect on. The
433 list can include specific status codes (e.g. '503') or the strings
434 '4xx' / '5xx'.
435 reconnect_delay_max
437 Sets the maximum delay in seconds after which to give up
438 reconnecting
439 mime_type
441 Export the MIME type.
442 http_version
444 Exports the HTTP response version number. Usually "1.0" or "1.1".
445 icy If set to 1 request ICY (SHOUTcast) metadata from the server. If
447 the server supports this, the metadata has to be retrieved by the
448 application by reading the icy_metadata_headers and
449 icy_metadata_packet options. The default is 1.
450 icy_metadata_headers
452 If the server supports ICY metadata, this contains the ICY-specific
453 HTTP reply headers, separated by newline characters.
454 icy_metadata_packet
456 If the server supports ICY metadata, and icy was set to 1, this
457 contains the last non-empty metadata packet sent by the server. It
458 should be polled in regular intervals by applications interested in
459 mid-stream metadata updates.
460 cookies
462 Set the cookies to be sent in future requests. The format of each
463 cookie is the same as the value of a Set-Cookie HTTP response
464 field. Multiple cookies can be delimited by a newline character.
465 offset
467 Set initial byte offset.
468 end_offset
470 Try to limit the request to bytes preceding this offset.
471 method
473 When used as a client option it sets the HTTP method for the
474 request.
475 When used as a server option it sets the HTTP method that is going
477 to be expected from the client(s). If the expected and the
478 received HTTP method do not match the client will be given a Bad
479 Request response. When unset the HTTP method is not checked for
480 now. This will be replaced by autodetection in the future.
481 listen
483 If set to 1 enables experimental HTTP server. This can be used to
484 send data when used as an output option, or read data from a client
485 with HTTP POST when used as an input option. If set to 2 enables
486 experimental multi-client HTTP server. This is not yet implemented
487 in ffmpeg.c and thus must not be used as a command line option.
488 # Server side (sending):
490 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>
491 # Client side (receiving):
493 ffmpeg -i http://<server>:<port> -c copy somefile.ogg
494 # Client can also be done with wget:
496 wget http://<server>:<port> -O somefile.ogg
497 # Server side (receiving):
499 ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg
500 # Client side (sending):
502 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>
503 # Client can also be done with wget:
505 wget --post-file=somefile.ogg http://<server>:<port>
506 send_expect_100
508 Send an Expect: 100-continue header for POST. If set to 1 it will
509 send, if set to 0 it won't, if set to -1 it will try to send if it
510 is applicable. Default value is -1.
511 auth_type
513 Set HTTP authentication type. No option for Digest, since this
514 method requires getting nonce parameters from the server first and
515 can't be used straight away like Basic.
516 none
518 Choose the HTTP authentication type automatically. This is the
519 default.
520 basic
522 Choose the HTTP basic authentication.
523 Basic authentication sends a Base64-encoded string that
525 contains a user name and password for the client. Base64 is not
526 a form of encryption and should be considered the same as
527 sending the user name and password in clear text (Base64 is a
528 reversible encoding). If a resource needs to be protected,
529 strongly consider using an authentication scheme other than
530 basic authentication. HTTPS/TLS should be used with basic
531 authentication. Without these additional security
532 enhancements, basic authentication should not be used to
533 protect sensitive or valuable information.
534 HTTP Cookies
536 Some HTTP requests will be denied unless cookie values are passed in
538 with the request. The cookies option allows these cookies to be
539 specified. At the very least, each cookie must specify a value along
540 with a path and domain. HTTP requests that match both the domain and
541 path will automatically include the cookie value in the HTTP Cookie
542 header field. Multiple cookies can be delimited by a newline.
543 The required syntax to play a stream specifying a cookie is:
545 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
547 Icecast
549 Icecast protocol (stream to Icecast servers)
550 This protocol accepts the following options:
552 ice_genre
554 Set the stream genre.
555 ice_name
557 Set the stream name.
558 ice_description
560 Set the stream description.
561 ice_url
563 Set the stream website URL.
564 ice_public
566 Set if the stream should be public. The default is 0 (not public).
567 user_agent
569 Override the User-Agent header. If not specified a string of the
570 form "Lavf/<version>" will be used.
571 password
573 Set the Icecast mountpoint password.
574 content_type
576 Set the stream content type. This must be set if it is different
577 from audio/mpeg.
578 legacy_icecast
580 This enables support for Icecast versions < 2.4.0, that do not
581 support the HTTP PUT method but the SOURCE method.
582 tls Establish a TLS (HTTPS) connection to Icecast.
584 icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>
586 ipfs
588 InterPlanetary File System (IPFS) protocol support. One can access
589 files stored on the IPFS network through so-called gateways. These are
590 http(s) endpoints. This protocol wraps the IPFS native protocols
591 (ipfs:// and ipns://) to be sent to such a gateway. Users can (and
592 should) host their own node which means this protocol will use one's
593 local gateway to access files on the IPFS network.
594 This protocol accepts the following options:
596 gateway
598 Defines the gateway to use. When not set, the protocol will first
599 try locating the local gateway by looking at $IPFS_GATEWAY,
600 $IPFS_PATH and "$HOME/.ipfs/", in that order.
601 One can use this protocol in 2 ways. Using IPFS:
603 ffplay ipfs://<hash>
605 Or the IPNS protocol (IPNS is mutable IPFS):
607 ffplay ipns://<hash>
609 mmst
611 MMS (Microsoft Media Server) protocol over TCP.
612 mmsh
614 MMS (Microsoft Media Server) protocol over HTTP.
615 The required syntax is:
617 mmsh://<server>[:<port>][/<app>][/<playpath>]
619 md5
621 MD5 output protocol.
622 Computes the MD5 hash of the data to be written, and on close writes
624 this to the designated output or stdout if none is specified. It can be
625 used to test muxers without writing an actual file.
626 Some examples follow.
628 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
630 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
631 # Write the MD5 hash of the encoded AVI file to stdout.
633 ffmpeg -i input.flv -f avi -y md5:
634 Note that some formats (typically MOV) require the output protocol to
636 be seekable, so they will fail with the MD5 output protocol.
637 pipe
639 UNIX pipe access protocol.
640 Read and write from UNIX pipes.
642 The accepted syntax is:
644 pipe:[<number>]
646 If fd isn't specified, number is the number corresponding to the file
648 descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).
649 If number is not specified, by default the stdout file descriptor will
650 be used for writing, stdin for reading.
651 For example to read from stdin with ffmpeg:
653 cat test.wav | ffmpeg -i pipe:0
655 # ...this is the same as...
656 cat test.wav | ffmpeg -i pipe:
657 For writing to stdout with ffmpeg:
659 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
661 # ...this is the same as...
662 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
663 This protocol accepts the following options:
665 blocksize
667 Set I/O operation maximum block size, in bytes. Default value is
668 "INT_MAX", which results in not limiting the requested block size.
669 Setting this value reasonably low improves user termination request
670 reaction time, which is valuable if data transmission is slow.
671 fd Set file descriptor.
673 Note that some formats (typically MOV), require the output protocol to
675 be seekable, so they will fail with the pipe output protocol.
676 prompeg
678 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
679 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction
681 mechanism for MPEG-2 Transport Streams sent over RTP.
682 This protocol must be used in conjunction with the "rtp_mpegts" muxer
684 and the "rtp" protocol.
685 The required syntax is:
687 -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>
689 The destination UDP ports are "port + 2" for the column FEC stream and
691 "port + 4" for the row FEC stream.
692 This protocol accepts the following options:
694 l=n The number of columns (4-20, LxD <= 100)
696 d=n The number of rows (4-20, LxD <= 100)
698 Example usage:
700 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>
702 rist
704 Reliable Internet Streaming Transport protocol
705 The accepted options are:
707 rist_profile
709 Supported values:
710 simple
712 main
713 This one is default.
714 advanced
716 buffer_size
717 Set internal RIST buffer size in milliseconds for retransmission of
718 data. Default value is 0 which means the librist default (1 sec).
719 Maximum value is 30 seconds.
720 fifo_size
722 Size of the librist receiver output fifo in number of packets. This
723 must be a power of 2. Defaults to 8192 (vs the librist default of
724 1024).
725 overrun_nonfatal=1|0
727 Survive in case of librist fifo buffer overrun. Default value is 0.
728 pkt_size
730 Set maximum packet size for sending data. 1316 by default.
731 log_level
733 Set loglevel for RIST logging messages. You only need to set this
734 if you explicitly want to enable debug level messages or packet
735 loss simulation, otherwise the regular loglevel is respected.
736 secret
738 Set override of encryption secret, by default is unset.
739 encryption
741 Set encryption type, by default is disabled. Acceptable values are
742 128 and 256.
743 rtmp
745 Real-Time Messaging Protocol.
746 The Real-Time Messaging Protocol (RTMP) is used for streaming
748 multimedia content across a TCP/IP network.
749 The required syntax is:
751 rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]
753 The accepted parameters are:
755 username
757 An optional username (mostly for publishing).
758 password
760 An optional password (mostly for publishing).
761 server
763 The address of the RTMP server.
764 port
766 The number of the TCP port to use (by default is 1935).
767 app It is the name of the application to access. It usually corresponds
769 to the path where the application is installed on the RTMP server
770 (e.g. /ondemand/, /flash/live/, etc.). You can override the value
771 parsed from the URI through the "rtmp_app" option, too.
772 playpath
774 It is the path or name of the resource to play with reference to
775 the application specified in app, may be prefixed by "mp4:". You
776 can override the value parsed from the URI through the
777 "rtmp_playpath" option, too.
778 listen
780 Act as a server, listening for an incoming connection.
781 timeout
783 Maximum time to wait for the incoming connection. Implies listen.
784 Additionally, the following parameters can be set via command line
786 options (or in code via "AVOption"s):
787 rtmp_app
789 Name of application to connect on the RTMP server. This option
790 overrides the parameter specified in the URI.
791 rtmp_buffer
793 Set the client buffer time in milliseconds. The default is 3000.
794 rtmp_conn
796 Extra arbitrary AMF connection parameters, parsed from a string,
797 e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0". Each
798 value is prefixed by a single character denoting the type, B for
799 Boolean, N for number, S for string, O for object, or Z for null,
800 followed by a colon. For Booleans the data must be either 0 or 1
801 for FALSE or TRUE, respectively. Likewise for Objects the data
802 must be 0 or 1 to end or begin an object, respectively. Data items
803 in subobjects may be named, by prefixing the type with 'N' and
804 specifying the name before the value (i.e. "NB:myFlag:1"). This
805 option may be used multiple times to construct arbitrary AMF
806 sequences.
807 rtmp_flashver
809 Version of the Flash plugin used to run the SWF player. The default
810 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0
811 (compatible; <libavformat version>).)
812 rtmp_flush_interval
814 Number of packets flushed in the same request (RTMPT only). The
815 default is 10.
816 rtmp_live
818 Specify that the media is a live stream. No resuming or seeking in
819 live streams is possible. The default value is "any", which means
820 the subscriber first tries to play the live stream specified in the
821 playpath. If a live stream of that name is not found, it plays the
822 recorded stream. The other possible values are "live" and
823 "recorded".
824 rtmp_pageurl
826 URL of the web page in which the media was embedded. By default no
827 value will be sent.
828 rtmp_playpath
830 Stream identifier to play or to publish. This option overrides the
831 parameter specified in the URI.
832 rtmp_subscribe
834 Name of live stream to subscribe to. By default no value will be
835 sent. It is only sent if the option is specified or if rtmp_live
836 is set to live.
837 rtmp_swfhash
839 SHA256 hash of the decompressed SWF file (32 bytes).
840 rtmp_swfsize
842 Size of the decompressed SWF file, required for SWFVerification.
843 rtmp_swfurl
845 URL of the SWF player for the media. By default no value will be
846 sent.
847 rtmp_swfverify
849 URL to player swf file, compute hash/size automatically.
850 rtmp_tcurl
852 URL of the target stream. Defaults to proto://host[:port]/app.
853 tcp_nodelay=1|0
855 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
856 Remark: Writing to the socket is currently not optimized to
858 minimize system calls and reduces the efficiency / effect of
859 TCP_NODELAY.
860 For example to read with ffplay a multimedia resource named "sample"
862 from the application "vod" from an RTMP server "myserver":
863 ffplay rtmp://myserver/vod/sample
865 To publish to a password protected server, passing the playpath and app
867 names separately:
868 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/
870 rtmpe
872 Encrypted Real-Time Messaging Protocol.
873 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
875 streaming multimedia content within standard cryptographic primitives,
876 consisting of Diffie-Hellman key exchange and HMACSHA256, generating a
877 pair of RC4 keys.
878 rtmps
880 Real-Time Messaging Protocol over a secure SSL connection.
881 The Real-Time Messaging Protocol (RTMPS) is used for streaming
883 multimedia content across an encrypted connection.
884 rtmpt
886 Real-Time Messaging Protocol tunneled through HTTP.
887 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
889 for streaming multimedia content within HTTP requests to traverse
890 firewalls.
891 rtmpte
893 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
894 The Encrypted Real-Time Messaging Protocol tunneled through HTTP
896 (RTMPTE) is used for streaming multimedia content within HTTP requests
897 to traverse firewalls.
898 rtmpts
900 Real-Time Messaging Protocol tunneled through HTTPS.
901 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is
903 used for streaming multimedia content within HTTPS requests to traverse
904 firewalls.
905 libsmbclient
907 libsmbclient permits one to manipulate CIFS/SMB network resources.
908 Following syntax is required.
910 smb://[[domain:]user[:password@]]server[/share[/path[/file]]]
912 This protocol accepts the following options.
914 timeout
916 Set timeout in milliseconds of socket I/O operations used by the
917 underlying low level operation. By default it is set to -1, which
918 means that the timeout is not specified.
919 truncate
921 Truncate existing files on write, if set to 1. A value of 0
922 prevents truncating. Default value is 1.
923 workgroup
925 Set the workgroup used for making connections. By default workgroup
926 is not specified.
927 For more information see: <http://www.samba.org/>.
929 libssh
931 Secure File Transfer Protocol via libssh
932 Read from or write to remote resources using SFTP protocol.
934 Following syntax is required.
936 sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
938 This protocol accepts the following options.
940 timeout
942 Set timeout of socket I/O operations used by the underlying low
943 level operation. By default it is set to -1, which means that the
944 timeout is not specified.
945 truncate
947 Truncate existing files on write, if set to 1. A value of 0
948 prevents truncating. Default value is 1.
949 private_key
951 Specify the path of the file containing private key to use during
952 authorization. By default libssh searches for keys in the ~/.ssh/
953 directory.
954 Example: Play a file stored on remote server.
956 ffplay sftp://user:password@server_address:22/home/user/resource.mpeg
958 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
960 Real-Time Messaging Protocol and its variants supported through
961 librtmp.
962 Requires the presence of the librtmp headers and library during
964 configuration. You need to explicitly configure the build with
965 "--enable-librtmp". If enabled this will replace the native RTMP
966 protocol.
967 This protocol provides most client functions and a few server functions
969 needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP
970 (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these
971 encrypted types (RTMPTE, RTMPTS).
972 The required syntax is:
974 <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>
976 where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe",
978 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
979 server, port, app and playpath have the same meaning as specified for
980 the RTMP native protocol. options contains a list of space-separated
981 options of the form key=val.
982 See the librtmp manual page (man 3 librtmp) for more information.
984 For example, to stream a file in real-time to an RTMP server using
986 ffmpeg:
987 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
989 To play the same stream using ffplay:
991 ffplay "rtmp://myserver/live/mystream live=1"
993 rtp
995 Real-time Transport Protocol.
996 The required syntax for an RTP URL is:
998 rtp://hostname[:port][?option=val...]
999 port specifies the RTP port to use.
1001 The following URL options are supported:
1003 ttl=n
1005 Set the TTL (Time-To-Live) value (for multicast only).
1006 rtcpport=n
1008 Set the remote RTCP port to n.
1009 localrtpport=n
1011 Set the local RTP port to n.
1012 localrtcpport=n'
1014 Set the local RTCP port to n.
1015 pkt_size=n
1017 Set max packet size (in bytes) to n.
1018 buffer_size=size
1020 Set the maximum UDP socket buffer size in bytes.
1021 connect=0|1
1023 Do a connect() on the UDP socket (if set to 1) or not (if set to
1024 0).
1025 sources=ip[,ip]
1027 List allowed source IP addresses.
1028 block=ip[,ip]
1030 List disallowed (blocked) source IP addresses.
1031 write_to_source=0|1
1033 Send packets to the source address of the latest received packet
1034 (if set to 1) or to a default remote address (if set to 0).
1035 localport=n
1037 Set the local RTP port to n.
1038 localaddr=addr
1040 Local IP address of a network interface used for sending packets or
1041 joining multicast groups.
1042 timeout=n
1044 Set timeout (in microseconds) of socket I/O operations to n.
1045 This is a deprecated option. Instead, localrtpport should be used.
1047 Important notes:
1049 1. If rtcpport is not set the RTCP port will be set to the RTP port
1051 value plus 1.
1052 2. If localrtpport (the local RTP port) is not set any available port
1054 will be used for the local RTP and RTCP ports.
1055 3. If localrtcpport (the local RTCP port) is not set it will be set to
1057 the local RTP port value plus 1.
1058 rtsp
1060 Real-Time Streaming Protocol.
1061 RTSP is not technically a protocol handler in libavformat, it is a
1063 demuxer and muxer. The demuxer supports both normal RTSP (with data
1064 transferred over RTP; this is used by e.g. Apple and Microsoft) and
1065 Real-RTSP (with data transferred over RDT).
1066 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
1068 supporting it (currently Darwin Streaming Server and Mischa
1069 Spiegelmock's <https://github.com/revmischa/rtsp-server>).
1070 The required syntax for a RTSP url is:
1072 rtsp://<hostname>[:<port>]/<path>
1074 Options can be set on the ffmpeg/ffplay command line, or set in code
1076 via "AVOption"s or in "avformat_open_input".
1077 Muxer
1079 The following options are supported.
1081 rtsp_transport
1083 Set RTSP transport protocols.
1084 It accepts the following values:
1086 udp Use UDP as lower transport protocol.
1088 tcp Use TCP (interleaving within the RTSP control channel) as lower
1090 transport protocol.
1091 Default value is 0.
1093 rtsp_flags
1095 Set RTSP flags.
1096 The following values are accepted:
1098 latm
1100 Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.
1101 rfc2190
1103 Use RFC 2190 packetization instead of RFC 4629 for H.263.
1104 skip_rtcp
1106 Don't send RTCP sender reports.
1107 h264_mode0
1109 Use mode 0 for H.264 in RTP.
1110 send_bye
1112 Send RTCP BYE packets when finishing.
1113 Default value is 0.
1115 min_port
1117 Set minimum local UDP port. Default value is 5000.
1118 max_port
1120 Set maximum local UDP port. Default value is 65000.
1121 buffer_size
1123 Set the maximum socket buffer size in bytes.
1124 pkt_size
1126 Set max send packet size (in bytes). Default value is 1472.
1127 Demuxer
1129 The following options are supported.
1131 initial_pause
1133 Do not start playing the stream immediately if set to 1. Default
1134 value is 0.
1135 rtsp_transport
1137 Set RTSP transport protocols.
1138 It accepts the following values:
1140 udp Use UDP as lower transport protocol.
1142 tcp Use TCP (interleaving within the RTSP control channel) as lower
1144 transport protocol.
1145 udp_multicast
1147 Use UDP multicast as lower transport protocol.
1148 http
1150 Use HTTP tunneling as lower transport protocol, which is useful
1151 for passing proxies.
1152 https
1154 Use HTTPs tunneling as lower transport protocol, which is
1155 useful for passing proxies and widely used for security
1156 consideration.
1157 Multiple lower transport protocols may be specified, in that case
1159 they are tried one at a time (if the setup of one fails, the next
1160 one is tried). For the muxer, only the tcp and udp options are
1161 supported.
1162 rtsp_flags
1164 Set RTSP flags.
1165 The following values are accepted:
1167 filter_src
1169 Accept packets only from negotiated peer address and port.
1170 listen
1172 Act as a server, listening for an incoming connection.
1173 prefer_tcp
1175 Try TCP for RTP transport first, if TCP is available as RTSP
1176 RTP transport.
1177 satip_raw
1179 Export raw MPEG-TS stream instead of demuxing. The flag will
1180 simply write out the raw stream, with the original PAT/PMT/PIDs
1181 intact.
1182 Default value is none.
1184 allowed_media_types
1186 Set media types to accept from the server.
1187 The following flags are accepted:
1189 video
1191 audio
1192 data
1193 subtitle
1194 By default it accepts all media types.
1196 min_port
1198 Set minimum local UDP port. Default value is 5000.
1199 max_port
1201 Set maximum local UDP port. Default value is 65000.
1202 listen_timeout
1204 Set maximum timeout (in seconds) to establish an initial
1205 connection. Setting listen_timeout > 0 sets rtsp_flags to listen.
1206 Default is -1 which means an infinite timeout when listen mode is
1207 set.
1208 reorder_queue_size
1210 Set number of packets to buffer for handling of reordered packets.
1211 timeout
1213 Set socket TCP I/O timeout in microseconds.
1214 user_agent
1216 Override User-Agent header. If not specified, it defaults to the
1217 libavformat identifier string.
1218 buffer_size
1220 Set the maximum socket buffer size in bytes.
1221 When receiving data over UDP, the demuxer tries to reorder received
1223 packets (since they may arrive out of order, or packets may get lost
1224 totally). This can be disabled by setting the maximum demuxing delay to
1225 zero (via the "max_delay" field of AVFormatContext).
1226 When watching multi-bitrate Real-RTSP streams with ffplay, the streams
1228 to display can be chosen with "-vst" n and "-ast" n for video and audio
1229 respectively, and can be switched on the fly by pressing "v" and "a".
1230 Examples
1232 The following examples all make use of the ffplay and ffmpeg tools.
1234 • Watch a stream over UDP, with a max reordering delay of 0.5
1236 seconds:
1237 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1239 • Watch a stream tunneled over HTTP:
1241 ffplay -rtsp_transport http rtsp://server/video.mp4
1243 • Send a stream in realtime to a RTSP server, for others to watch:
1245 ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1247 • Receive a stream in realtime:
1249 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>
1251 sap
1253 Session Announcement Protocol (RFC 2974). This is not technically a
1254 protocol handler in libavformat, it is a muxer and demuxer. It is used
1255 for signalling of RTP streams, by announcing the SDP for the streams
1256 regularly on a separate port.
1257 Muxer
1259 The syntax for a SAP url given to the muxer is:
1261 sap://<destination>[:<port>][?<options>]
1263 The RTP packets are sent to destination on port port, or to port 5004
1265 if no port is specified. options is a "&"-separated list. The
1266 following options are supported:
1267 announce_addr=address
1269 Specify the destination IP address for sending the announcements
1270 to. If omitted, the announcements are sent to the commonly used
1271 SAP announcement multicast address 224.2.127.254 (sap.mcast.net),
1272 or ff0e::2:7ffe if destination is an IPv6 address.
1273 announce_port=port
1275 Specify the port to send the announcements on, defaults to 9875 if
1276 not specified.
1277 ttl=ttl
1279 Specify the time to live value for the announcements and RTP
1280 packets, defaults to 255.
1281 same_port=0|1
1283 If set to 1, send all RTP streams on the same port pair. If zero
1284 (the default), all streams are sent on unique ports, with each
1285 stream on a port 2 numbers higher than the previous. VLC/Live555
1286 requires this to be set to 1, to be able to receive the stream.
1287 The RTP stack in libavformat for receiving requires all streams to
1288 be sent on unique ports.
1289 Example command lines follow.
1291 To broadcast a stream on the local subnet, for watching in VLC:
1293 ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1
1295 Similarly, for watching in ffplay:
1297 ffmpeg -re -i <input> -f sap sap://224.0.0.255
1299 And for watching in ffplay, over IPv6:
1301 ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]
1303 Demuxer
1305 The syntax for a SAP url given to the demuxer is:
1307 sap://[<address>][:<port>]
1309 address is the multicast address to listen for announcements on, if
1311 omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the
1312 port that is listened on, 9875 if omitted.
1313 The demuxers listens for announcements on the given address and port.
1315 Once an announcement is received, it tries to receive that particular
1316 stream.
1317 Example command lines follow.
1319 To play back the first stream announced on the normal SAP multicast
1321 address:
1322 ffplay sap://
1324 To play back the first stream announced on one the default IPv6 SAP
1326 multicast address:
1327 ffplay sap://[ff0e::2:7ffe]
1329 sctp
1331 Stream Control Transmission Protocol.
1332 The accepted URL syntax is:
1334 sctp://<host>:<port>[?<options>]
1336 The protocol accepts the following options:
1338 listen
1340 If set to any value, listen for an incoming connection. Outgoing
1341 connection is done by default.
1342 max_streams
1344 Set the maximum number of streams. By default no limit is set.
1345 srt
1347 Haivision Secure Reliable Transport Protocol via libsrt.
1348 The supported syntax for a SRT URL is:
1350 srt://<hostname>:<port>[?<options>]
1352 options contains a list of &-separated options of the form key=val.
1354 or
1356 <options> srt://<hostname>:<port>
1358 options contains a list of '-key val' options.
1360 This protocol accepts the following options.
1362 connect_timeout=milliseconds
1364 Connection timeout; SRT cannot connect for RTT > 1500 msec (2
1365 handshake exchanges) with the default connect timeout of 3 seconds.
1366 This option applies to the caller and rendezvous connection modes.
1367 The connect timeout is 10 times the value set for the rendezvous
1368 mode (which can be used as a workaround for this connection problem
1369 with earlier versions).
1370 ffs=bytes
1372 Flight Flag Size (Window Size), in bytes. FFS is actually an
1373 internal parameter and you should set it to not less than
1374 recv_buffer_size and mss. The default value is relatively large,
1375 therefore unless you set a very large receiver buffer, you do not
1376 need to change this option. Default value is 25600.
1377 inputbw=bytes/seconds
1379 Sender nominal input rate, in bytes per seconds. Used along with
1380 oheadbw, when maxbw is set to relative (0), to calculate maximum
1381 sending rate when recovery packets are sent along with the main
1382 media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set
1383 while maxbw is set to relative (0), the actual input rate is
1384 evaluated inside the library. Default value is 0.
1385 iptos=tos
1387 IP Type of Service. Applies to sender only. Default value is 0xB8.
1388 ipttl=ttl
1390 IP Time To Live. Applies to sender only. Default value is 64.
1391 latency=microseconds
1393 Timestamp-based Packet Delivery Delay. Used to absorb bursts of
1394 missed packet retransmissions. This flag sets both rcvlatency and
1395 peerlatency to the same value. Note that prior to version 1.3.0
1396 this is the only flag to set the latency, however this is
1397 effectively equivalent to setting peerlatency, when side is sender
1398 and rcvlatency when side is receiver, and the bidirectional stream
1399 sending is not supported.
1400 listen_timeout=microseconds
1402 Set socket listen timeout.
1403 maxbw=bytes/seconds
1405 Maximum sending bandwidth, in bytes per seconds. -1 infinite
1406 (CSRTCC limit is 30mbps) 0 relative to input rate (see inputbw) >0
1407 absolute limit value Default value is 0 (relative)
1408 mode=caller|listener|rendezvous
1410 Connection mode. caller opens client connection. listener starts
1411 server to listen for incoming connections. rendezvous use Rendez-
1412 Vous connection mode. Default value is caller.
1413 mss=bytes
1415 Maximum Segment Size, in bytes. Used for buffer allocation and rate
1416 calculation using a packet counter assuming fully filled packets.
1417 The smallest MSS between the peers is used. This is 1500 by default
1418 in the overall internet. This is the maximum size of the UDP
1419 packet and can be only decreased, unless you have some unusual
1420 dedicated network settings. Default value is 1500.
1421 nakreport=1|0
1423 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1424 periodically until a lost packet is retransmitted or intentionally
1425 dropped. Default value is 1.
1426 oheadbw=percents
1428 Recovery bandwidth overhead above input rate, in percents. See
1429 inputbw. Default value is 25%.
1430 passphrase=string
1432 HaiCrypt Encryption/Decryption Passphrase string, length from 10 to
1433 79 characters. The passphrase is the shared secret between the
1434 sender and the receiver. It is used to generate the Key Encrypting
1435 Key using PBKDF2 (Password-Based Key Derivation Function). It is
1436 used only if pbkeylen is non-zero. It is used on the receiver only
1437 if the received data is encrypted. The configured passphrase
1438 cannot be recovered (write-only).
1439 enforced_encryption=1|0
1441 If true, both connection parties must have the same password set
1442 (including empty, that is, with no encryption). If the password
1443 doesn't match or only one side is unencrypted, the connection is
1444 rejected. Default is true.
1445 kmrefreshrate=packets
1447 The number of packets to be transmitted after which the encryption
1448 key is switched to a new key. Default is -1. -1 means auto
1449 (0x1000000 in srt library). The range for this option is integers
1450 in the 0 - "INT_MAX".
1451 kmpreannounce=packets
1453 The interval between when a new encryption key is sent and when
1454 switchover occurs. This value also applies to the subsequent
1455 interval between when switchover occurs and when the old encryption
1456 key is decommissioned. Default is -1. -1 means auto (0x1000 in srt
1457 library). The range for this option is integers in the 0 -
1458 "INT_MAX".
1459 snddropdelay=microseconds
1461 The sender's extra delay before dropping packets. This delay is
1462 added to the default drop delay time interval value.
1463 Special value -1: Do not drop packets on the sender at all.
1465 payload_size=bytes
1467 Sets the maximum declared size of a packet transferred during the
1468 single call to the sending function in Live mode. Use 0 if this
1469 value isn't used (which is default in file mode). Default is -1
1470 (automatic), which typically means MPEG-TS; if you are going to use
1471 SRT to send any different kind of payload, such as, for example,
1472 wrapping a live stream in very small frames, then you can use a
1473 bigger maximum frame size, though not greater than 1456 bytes.
1474 pkt_size=bytes
1476 Alias for payload_size.
1477 peerlatency=microseconds
1479 The latency value (as described in rcvlatency) that is set by the
1480 sender side as a minimum value for the receiver.
1481 pbkeylen=bytes
1483 Sender encryption key length, in bytes. Only can be set to 0, 16,
1484 24 and 32. Enable sender encryption if not 0. Not required on
1485 receiver (set to 0), key size obtained from sender in HaiCrypt
1486 handshake. Default value is 0.
1487 rcvlatency=microseconds
1489 The time that should elapse since the moment when the packet was
1490 sent and the moment when it's delivered to the receiver application
1491 in the receiving function. This time should be a buffer time large
1492 enough to cover the time spent for sending, unexpectedly extended
1493 RTT time, and the time needed to retransmit the lost UDP packet.
1494 The effective latency value will be the maximum of this options'
1495 value and the value of peerlatency set by the peer side. Before
1496 version 1.3.0 this option is only available as latency.
1497 recv_buffer_size=bytes
1499 Set UDP receive buffer size, expressed in bytes.
1500 send_buffer_size=bytes
1502 Set UDP send buffer size, expressed in bytes.
1503 timeout=microseconds
1505 Set raise error timeouts for read, write and connect operations.
1506 Note that the SRT library has internal timeouts which can be
1507 controlled separately, the value set here is only a cap on those.
1508 tlpktdrop=1|0
1510 Too-late Packet Drop. When enabled on receiver, it skips missing
1511 packets that have not been delivered in time and delivers the
1512 following packets to the application when their time-to-play has
1513 come. It also sends a fake ACK to the sender. When enabled on
1514 sender and enabled on the receiving peer, the sender drops the
1515 older packets that have no chance of being delivered in time. It
1516 was automatically enabled in the sender if the receiver supports
1517 it.
1518 sndbuf=bytes
1520 Set send buffer size, expressed in bytes.
1521 rcvbuf=bytes
1523 Set receive buffer size, expressed in bytes.
1524 Receive buffer must not be greater than ffs.
1526 lossmaxttl=packets
1528 The value up to which the Reorder Tolerance may grow. When Reorder
1529 Tolerance is > 0, then packet loss report is delayed until that
1530 number of packets come in. Reorder Tolerance increases every time a
1531 "belated" packet has come, but it wasn't due to retransmission
1532 (that is, when UDP packets tend to come out of order), with the
1533 difference between the latest sequence and this packet's sequence,
1534 and not more than the value of this option. By default it's 0,
1535 which means that this mechanism is turned off, and the loss report
1536 is always sent immediately upon experiencing a "gap" in sequences.
1537 minversion
1539 The minimum SRT version that is required from the peer. A
1540 connection to a peer that does not satisfy the minimum version
1541 requirement will be rejected.
1542 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1544 form.
1545 streamid=string
1547 A string limited to 512 characters that can be set on the socket
1548 prior to connecting. This stream ID will be able to be retrieved by
1549 the listener side from the socket that is returned from srt_accept
1550 and was connected by a socket with that set stream ID. SRT does not
1551 enforce any special interpretation of the contents of this string.
1552 This option doesn’t make sense in Rendezvous connection; the result
1553 might be that simply one side will override the value from the
1554 other side and it’s the matter of luck which one would win
1555 srt_streamid=string
1557 Alias for streamid to avoid conflict with ffmpeg command line
1558 option.
1559 smoother=live|file
1561 The type of Smoother used for the transmission for that socket,
1562 which is responsible for the transmission and congestion control.
1563 The Smoother type must be exactly the same on both connecting
1564 parties, otherwise the connection is rejected.
1565 messageapi=1|0
1567 When set, this socket uses the Message API, otherwise it uses
1568 Buffer API. Note that in live mode (see transtype) there’s only
1569 message API available. In File mode you can chose to use one of two
1570 modes:
1571 Stream API (default, when this option is false). In this mode you
1573 may send as many data as you wish with one sending instruction, or
1574 even use dedicated functions that read directly from a file. The
1575 internal facility will take care of any speed and congestion
1576 control. When receiving, you can also receive as many data as
1577 desired, the data not extracted will be waiting for the next call.
1578 There is no boundary between data portions in the Stream mode.
1579 Message API. In this mode your single sending instruction passes
1581 exactly one piece of data that has boundaries (a message). Contrary
1582 to Live mode, this message may span across multiple UDP packets and
1583 the only size limitation is that it shall fit as a whole in the
1584 sending buffer. The receiver shall use as large buffer as necessary
1585 to receive the message, otherwise the message will not be given up.
1586 When the message is not complete (not all packets received or there
1587 was a packet loss) it will not be given up.
1588 transtype=live|file
1590 Sets the transmission type for the socket, in particular, setting
1591 this option sets multiple other parameters to their default values
1592 as required for a particular transmission type.
1593 live: Set options as for live transmission. In this mode, you
1595 should send by one sending instruction only so many data that fit
1596 in one UDP packet, and limited to the value defined first in
1597 payload_size (1316 is default in this mode). There is no speed
1598 control in this mode, only the bandwidth control, if configured, in
1599 order to not exceed the bandwidth with the overhead transmission
1600 (retransmitted and control packets).
1601 file: Set options as for non-live transmission. See messageapi for
1603 further explanations
1604 linger=seconds
1606 The number of seconds that the socket waits for unsent data when
1607 closing. Default is -1. -1 means auto (off with 0 seconds in live
1608 mode, on with 180 seconds in file mode). The range for this option
1609 is integers in the 0 - "INT_MAX".
1610 tsbpd=1|0
1612 When true, use Timestamp-based Packet Delivery mode. The default
1613 behavior depends on the transmission type: enabled in live mode,
1614 disabled in file mode.
1615 For more information see: <https://github.com/Haivision/srt>.
1617 srtp
1619 Secure Real-time Transport Protocol.
1620 The accepted options are:
1622 srtp_in_suite
1624 srtp_out_suite
1625 Select input and output encoding suites.
1626 Supported values:
1628 AES_CM_128_HMAC_SHA1_80
1630 SRTP_AES128_CM_HMAC_SHA1_80
1631 AES_CM_128_HMAC_SHA1_32
1632 SRTP_AES128_CM_HMAC_SHA1_32
1633 srtp_in_params
1634 srtp_out_params
1635 Set input and output encoding parameters, which are expressed by a
1636 base64-encoded representation of a binary block. The first 16 bytes
1637 of this binary block are used as master key, the following 14 bytes
1638 are used as master salt.
1639 subfile
1641 Virtually extract a segment of a file or another stream. The
1642 underlying stream must be seekable.
1643 Accepted options:
1645 start
1647 Start offset of the extracted segment, in bytes.
1648 end End offset of the extracted segment, in bytes. If set to 0,
1650 extract till end of file.
1651 Examples:
1653 Extract a chapter from a DVD VOB file (start and end sectors obtained
1655 externally and multiplied by 2048):
1656 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1658 Play an AVI file directly from a TAR archive:
1660 subfile,,start,183241728,end,366490624,,:archive.tar
1662 Play a MPEG-TS file from start offset till end:
1664 subfile,,start,32815239,end,0,,:video.ts
1666 tee
1668 Writes the output to multiple protocols. The individual outputs are
1669 separated by |
1670 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1672 tcp
1674 Transmission Control Protocol.
1675 The required syntax for a TCP url is:
1677 tcp://<hostname>:<port>[?<options>]
1679 options contains a list of &-separated options of the form key=val.
1681 The list of supported options follows.
1683 listen=2|1|0
1685 Listen for an incoming connection. 0 disables listen, 1 enables
1686 listen in single client mode, 2 enables listen in multi-client
1687 mode. Default value is 0.
1688 timeout=microseconds
1690 Set raise error timeout, expressed in microseconds.
1691 This option is only relevant in read mode: if no data arrived in
1693 more than this time interval, raise error.
1694 listen_timeout=milliseconds
1696 Set listen timeout, expressed in milliseconds.
1697 recv_buffer_size=bytes
1699 Set receive buffer size, expressed bytes.
1700 send_buffer_size=bytes
1702 Set send buffer size, expressed bytes.
1703 tcp_nodelay=1|0
1705 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1706 Remark: Writing to the socket is currently not optimized to
1708 minimize system calls and reduces the efficiency / effect of
1709 TCP_NODELAY.
1710 tcp_mss=bytes
1712 Set maximum segment size for outgoing TCP packets, expressed in
1713 bytes.
1714 The following example shows how to setup a listening TCP connection
1716 with ffmpeg, which is then accessed with ffplay:
1717 ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
1719 ffplay tcp://<hostname>:<port>
1720 tls
1722 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1723 The required syntax for a TLS/SSL url is:
1725 tls://<hostname>:<port>[?<options>]
1727 The following parameters can be set via command line options (or in
1729 code via "AVOption"s):
1730 ca_file, cafile=filename
1732 A file containing certificate authority (CA) root certificates to
1733 treat as trusted. If the linked TLS library contains a default this
1734 might not need to be specified for verification to work, but not
1735 all libraries and setups have defaults built in. The file must be
1736 in OpenSSL PEM format.
1737 tls_verify=1|0
1739 If enabled, try to verify the peer that we are communicating with.
1740 Note, if using OpenSSL, this currently only makes sure that the
1741 peer certificate is signed by one of the root certificates in the
1742 CA database, but it does not validate that the certificate actually
1743 matches the host name we are trying to connect to. (With other
1744 backends, the host name is validated as well.)
1745 This is disabled by default since it requires a CA database to be
1747 provided by the caller in many cases.
1748 cert_file, cert=filename
1750 A file containing a certificate to use in the handshake with the
1751 peer. (When operating as server, in listen mode, this is more
1752 often required by the peer, while client certificates only are
1753 mandated in certain setups.)
1754 key_file, key=filename
1756 A file containing the private key for the certificate.
1757 listen=1|0
1759 If enabled, listen for connections on the provided port, and assume
1760 the server role in the handshake instead of the client role.
1761 http_proxy
1763 The HTTP proxy to tunnel through, e.g. "http://example.com:1234".
1764 The proxy must support the CONNECT method.
1765 Example command lines:
1767 To create a TLS/SSL server that serves an input stream.
1769 ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>
1771 To play back a stream from the TLS/SSL server using ffplay:
1773 ffplay tls://<hostname>:<port>
1775 udp
1777 User Datagram Protocol.
1778 The required syntax for an UDP URL is:
1780 udp://<hostname>:<port>[?<options>]
1782 options contains a list of &-separated options of the form key=val.
1784 In case threading is enabled on the system, a circular buffer is used
1786 to store the incoming data, which allows one to reduce loss of data due
1787 to UDP socket buffer overruns. The fifo_size and overrun_nonfatal
1788 options are related to this buffer.
1789 The list of supported options follows.
1791 buffer_size=size
1793 Set the UDP maximum socket buffer size in bytes. This is used to
1794 set either the receive or send buffer size, depending on what the
1795 socket is used for. Default is 32 KB for output, 384 KB for input.
1796 See also fifo_size.
1797 bitrate=bitrate
1799 If set to nonzero, the output will have the specified constant
1800 bitrate if the input has enough packets to sustain it.
1801 burst_bits=bits
1803 When using bitrate this specifies the maximum number of bits in
1804 packet bursts.
1805 localport=port
1807 Override the local UDP port to bind with.
1808 localaddr=addr
1810 Local IP address of a network interface used for sending packets or
1811 joining multicast groups.
1812 pkt_size=size
1814 Set the size in bytes of UDP packets.
1815 reuse=1|0
1817 Explicitly allow or disallow reusing UDP sockets.
1818 ttl=ttl
1820 Set the time to live value (for multicast only).
1821 connect=1|0
1823 Initialize the UDP socket with connect(). In this case, the
1824 destination address can't be changed with ff_udp_set_remote_url
1825 later. If the destination address isn't known at the start, this
1826 option can be specified in ff_udp_set_remote_url, too. This allows
1827 finding out the source address for the packets with getsockname,
1828 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1829 unreachable" is received. For receiving, this gives the benefit of
1830 only receiving packets from the specified peer address/port.
1831 sources=address[,address]
1833 Only receive packets sent from the specified addresses. In case of
1834 multicast, also subscribe to multicast traffic coming from these
1835 addresses only.
1836 block=address[,address]
1838 Ignore packets sent from the specified addresses. In case of
1839 multicast, also exclude the source addresses in the multicast
1840 subscription.
1841 fifo_size=units
1843 Set the UDP receiving circular buffer size, expressed as a number
1844 of packets with size of 188 bytes. If not specified defaults to
1845 7*4096.
1846 overrun_nonfatal=1|0
1848 Survive in case of UDP receiving circular buffer overrun. Default
1849 value is 0.
1850 timeout=microseconds
1852 Set raise error timeout, expressed in microseconds.
1853 This option is only relevant in read mode: if no data arrived in
1855 more than this time interval, raise error.
1856 broadcast=1|0
1858 Explicitly allow or disallow UDP broadcasting.
1859 Note that broadcasting may not work properly on networks having a
1861 broadcast storm protection.
1862 Examples
1864 • Use ffmpeg to stream over UDP to a remote endpoint:
1866 ffmpeg -i <input> -f <format> udp://<hostname>:<port>
1868 • Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP
1870 packets, using a large input buffer:
1871 ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535
1873 • Use ffmpeg to receive over UDP from a remote endpoint:
1875 ffmpeg -i udp://[<multicast-address>]:<port> ...
1877 unix
1879 Unix local socket
1880 The required syntax for a Unix socket URL is:
1882 unix://<filepath>
1884 The following parameters can be set via command line options (or in
1886 code via "AVOption"s):
1887 timeout
1889 Timeout in ms.
1890 listen
1892 Create the Unix socket in listening mode.
1893 zmq
1895 ZeroMQ asynchronous messaging using the libzmq library.
1896 This library supports unicast streaming to multiple clients without
1898 relying on an external server.
1899 The required syntax for streaming or connecting to a stream is:
1901 zmq:tcp://ip-address:port
1903 Example: Create a localhost stream on port 5555:
1905 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1907 Multiple clients may connect to the stream using:
1909 ffplay zmq:tcp://127.0.0.1:5555
1911 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub
1913 pattern. The server side binds to a port and publishes data. Clients
1914 connect to the server (via IP address/port) and subscribe to the
1915 stream. The order in which the server and client start generally does
1916 not matter.
1917 ffmpeg must be compiled with the --enable-libzmq option to support this
1919 protocol.
1920 Options can be set on the ffmpeg/ffplay command line. The following
1922 options are supported:
1923 pkt_size
1925 Forces the maximum packet size for sending/receiving data. The
1926 default value is 131,072 bytes. On the server side, this sets the
1927 maximum size of sent packets via ZeroMQ. On the clients, it sets an
1928 internal buffer size for receiving packets. Note that pkt_size on
1929 the clients should be equal to or greater than pkt_size on the
1930 server. Otherwise the received message may be truncated causing
1931 decoding errors.
1932
1934 ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
1935
1937 The FFmpeg developers.
1938 For details about the authorship, see the Git history of the project
1940 (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
1941 the FFmpeg source directory, or browsing the online repository at
1942 <https://git.ffmpeg.org/ffmpeg>.
1943 Maintainers for the specific components are listed in the file
1945 MAINTAINERS in the source code tree.
1946 FFMPEG-PROTOCOLS(1)