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 file
250 File access protocol.
251 Read from or write to a file.
253 A file URL can have the form:
255 file:<filename>
257 where filename is the path of the file to read.
259 An URL that does not have a protocol prefix will be assumed to be a
261 file URL. Depending on the build, an URL that looks like a Windows path
262 with the drive letter at the beginning will also be assumed to be a
263 file URL (usually not the case in builds for unix-like systems).
264 For example to read from a file input.mpeg with ffmpeg use the command:
266 ffmpeg -i file:input.mpeg output.mpeg
268 This protocol accepts the following options:
270 truncate
272 Truncate existing files on write, if set to 1. A value of 0
273 prevents truncating. Default value is 1.
274 blocksize
276 Set I/O operation maximum block size, in bytes. Default value is
277 "INT_MAX", which results in not limiting the requested block size.
278 Setting this value reasonably low improves user termination request
279 reaction time, which is valuable for files on slow medium.
280 follow
282 If set to 1, the protocol will retry reading at the end of the
283 file, allowing reading files that still are being written. In order
284 for this to terminate, you either need to use the rw_timeout
285 option, or use the interrupt callback (for API users).
286 seekable
288 Controls if seekability is advertised on the file. 0 means non-
289 seekable, -1 means auto (seekable for normal files, non-seekable
290 for named pipes).
291 Many demuxers handle seekable and non-seekable resources
293 differently, overriding this might speed up opening certain files
294 at the cost of losing some features (e.g. accurate seeking).
295 ftp
297 FTP (File Transfer Protocol).
298 Read from or write to remote resources using FTP protocol.
300 Following syntax is required.
302 ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
304 This protocol accepts the following options.
306 timeout
308 Set timeout in microseconds of socket I/O operations used by the
309 underlying low level operation. By default it is set to -1, which
310 means that the timeout is not specified.
311 ftp-user
313 Set a user to be used for authenticating to the FTP server. This is
314 overridden by the user in the FTP URL.
315 ftp-password
317 Set a password to be used for authenticating to the FTP server.
318 This is overridden by the password in the FTP URL, or by ftp-
319 anonymous-password if no user is set.
320 ftp-anonymous-password
322 Password used when login as anonymous user. Typically an e-mail
323 address should be used.
324 ftp-write-seekable
326 Control seekability of connection during encoding. If set to 1 the
327 resource is supposed to be seekable, if set to 0 it is assumed not
328 to be seekable. Default value is 0.
329 NOTE: Protocol can be used as output, but it is recommended to not do
331 it, unless special care is taken (tests, customized server
332 configuration etc.). Different FTP servers behave in different way
333 during seek operation. ff* tools may produce incomplete content due to
334 server limitations.
335 gopher
337 Gopher protocol.
338 gophers
340 Gophers protocol.
341 The Gopher protocol with TLS encapsulation.
343 hls
345 Read Apple HTTP Live Streaming compliant segmented stream as a uniform
346 one. The M3U8 playlists describing the segments can be remote HTTP
347 resources or local files, accessed using the standard file protocol.
348 The nested protocol is declared by specifying "+proto" after the hls
349 URI scheme name, where proto is either "file" or "http".
350 hls+http://host/path/to/remote/resource.m3u8
352 hls+file://path/to/local/resource.m3u8
353 Using this protocol is discouraged - the hls demuxer should work just
355 as well (if not, please report the issues) and is more complete. To
356 use the hls demuxer instead, simply use the direct URLs to the m3u8
357 files.
358 http
360 HTTP (Hyper Text Transfer Protocol).
361 This protocol accepts the following options:
363 seekable
365 Control seekability of connection. If set to 1 the resource is
366 supposed to be seekable, if set to 0 it is assumed not to be
367 seekable, if set to -1 it will try to autodetect if it is seekable.
368 Default value is -1.
369 chunked_post
371 If set to 1 use chunked Transfer-Encoding for posts, default is 1.
372 content_type
374 Set a specific content type for the POST messages or for listen
375 mode.
376 http_proxy
378 set HTTP proxy to tunnel through e.g. http://example.com:1234
379 headers
381 Set custom HTTP headers, can override built in default headers. The
382 value must be a string encoding the headers.
383 multiple_requests
385 Use persistent connections if set to 1, default is 0.
386 post_data
388 Set custom HTTP post data.
389 referer
391 Set the Referer header. Include 'Referer: URL' header in HTTP
392 request.
393 user_agent
395 Override the User-Agent header. If not specified the protocol will
396 use a string describing the libavformat build. ("Lavf/<version>")
397 reconnect_at_eof
399 If set then eof is treated like an error and causes reconnection,
400 this is useful for live / endless streams.
401 reconnect_streamed
403 If set then even streamed/non seekable streams will be reconnected
404 on errors.
405 reconnect_on_network_error
407 Reconnect automatically in case of TCP/TLS errors during connect.
408 reconnect_on_http_error
410 A comma separated list of HTTP status codes to reconnect on. The
411 list can include specific status codes (e.g. '503') or the strings
412 '4xx' / '5xx'.
413 reconnect_delay_max
415 Sets the maximum delay in seconds after which to give up
416 reconnecting
417 mime_type
419 Export the MIME type.
420 http_version
422 Exports the HTTP response version number. Usually "1.0" or "1.1".
423 icy If set to 1 request ICY (SHOUTcast) metadata from the server. If
425 the server supports this, the metadata has to be retrieved by the
426 application by reading the icy_metadata_headers and
427 icy_metadata_packet options. The default is 1.
428 icy_metadata_headers
430 If the server supports ICY metadata, this contains the ICY-specific
431 HTTP reply headers, separated by newline characters.
432 icy_metadata_packet
434 If the server supports ICY metadata, and icy was set to 1, this
435 contains the last non-empty metadata packet sent by the server. It
436 should be polled in regular intervals by applications interested in
437 mid-stream metadata updates.
438 cookies
440 Set the cookies to be sent in future requests. The format of each
441 cookie is the same as the value of a Set-Cookie HTTP response
442 field. Multiple cookies can be delimited by a newline character.
443 offset
445 Set initial byte offset.
446 end_offset
448 Try to limit the request to bytes preceding this offset.
449 method
451 When used as a client option it sets the HTTP method for the
452 request.
453 When used as a server option it sets the HTTP method that is going
455 to be expected from the client(s). If the expected and the
456 received HTTP method do not match the client will be given a Bad
457 Request response. When unset the HTTP method is not checked for
458 now. This will be replaced by autodetection in the future.
459 listen
461 If set to 1 enables experimental HTTP server. This can be used to
462 send data when used as an output option, or read data from a client
463 with HTTP POST when used as an input option. If set to 2 enables
464 experimental multi-client HTTP server. This is not yet implemented
465 in ffmpeg.c and thus must not be used as a command line option.
466 # Server side (sending):
468 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>
469 # Client side (receiving):
471 ffmpeg -i http://<server>:<port> -c copy somefile.ogg
472 # Client can also be done with wget:
474 wget http://<server>:<port> -O somefile.ogg
475 # Server side (receiving):
477 ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg
478 # Client side (sending):
480 ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>
481 # Client can also be done with wget:
483 wget --post-file=somefile.ogg http://<server>:<port>
484 send_expect_100
486 Send an Expect: 100-continue header for POST. If set to 1 it will
487 send, if set to 0 it won't, if set to -1 it will try to send if it
488 is applicable. Default value is -1.
489 auth_type
491 Set HTTP authentication type. No option for Digest, since this
492 method requires getting nonce parameters from the server first and
493 can't be used straight away like Basic.
494 none
496 Choose the HTTP authentication type automatically. This is the
497 default.
498 basic
500 Choose the HTTP basic authentication.
501 Basic authentication sends a Base64-encoded string that
503 contains a user name and password for the client. Base64 is not
504 a form of encryption and should be considered the same as
505 sending the user name and password in clear text (Base64 is a
506 reversible encoding). If a resource needs to be protected,
507 strongly consider using an authentication scheme other than
508 basic authentication. HTTPS/TLS should be used with basic
509 authentication. Without these additional security
510 enhancements, basic authentication should not be used to
511 protect sensitive or valuable information.
512 HTTP Cookies
514 Some HTTP requests will be denied unless cookie values are passed in
516 with the request. The cookies option allows these cookies to be
517 specified. At the very least, each cookie must specify a value along
518 with a path and domain. HTTP requests that match both the domain and
519 path will automatically include the cookie value in the HTTP Cookie
520 header field. Multiple cookies can be delimited by a newline.
521 The required syntax to play a stream specifying a cookie is:
523 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
525 Icecast
527 Icecast protocol (stream to Icecast servers)
528 This protocol accepts the following options:
530 ice_genre
532 Set the stream genre.
533 ice_name
535 Set the stream name.
536 ice_description
538 Set the stream description.
539 ice_url
541 Set the stream website URL.
542 ice_public
544 Set if the stream should be public. The default is 0 (not public).
545 user_agent
547 Override the User-Agent header. If not specified a string of the
548 form "Lavf/<version>" will be used.
549 password
551 Set the Icecast mountpoint password.
552 content_type
554 Set the stream content type. This must be set if it is different
555 from audio/mpeg.
556 legacy_icecast
558 This enables support for Icecast versions < 2.4.0, that do not
559 support the HTTP PUT method but the SOURCE method.
560 tls Establish a TLS (HTTPS) connection to Icecast.
562 icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>
564 ipfs
566 InterPlanetary File System (IPFS) protocol support. One can access
567 files stored on the IPFS network through so-called gateways. These are
568 http(s) endpoints. This protocol wraps the IPFS native protocols
569 (ipfs:// and ipns://) to be sent to such a gateway. Users can (and
570 should) host their own node which means this protocol will use one's
571 local gateway to access files on the IPFS network.
572 If a user doesn't have a node of their own then the public gateway
574 "https://dweb.link" is used by default.
575 This protocol accepts the following options:
577 gateway
579 Defines the gateway to use. When not set, the protocol will first
580 try locating the local gateway by looking at $IPFS_GATEWAY,
581 $IPFS_PATH and "$HOME/.ipfs/", in that order. If that fails
582 "https://dweb.link" will be used.
583 One can use this protocol in 2 ways. Using IPFS:
585 ffplay ipfs://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T
587 Or the IPNS protocol (IPNS is mutable IPFS):
589 ffplay ipns://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T
591 mmst
593 MMS (Microsoft Media Server) protocol over TCP.
594 mmsh
596 MMS (Microsoft Media Server) protocol over HTTP.
597 The required syntax is:
599 mmsh://<server>[:<port>][/<app>][/<playpath>]
601 md5
603 MD5 output protocol.
604 Computes the MD5 hash of the data to be written, and on close writes
606 this to the designated output or stdout if none is specified. It can be
607 used to test muxers without writing an actual file.
608 Some examples follow.
610 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
612 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
613 # Write the MD5 hash of the encoded AVI file to stdout.
615 ffmpeg -i input.flv -f avi -y md5:
616 Note that some formats (typically MOV) require the output protocol to
618 be seekable, so they will fail with the MD5 output protocol.
619 pipe
621 UNIX pipe access protocol.
622 Read and write from UNIX pipes.
624 The accepted syntax is:
626 pipe:[<number>]
628 number is the number corresponding to the file descriptor of the pipe
630 (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not
631 specified, by default the stdout file descriptor will be used for
632 writing, stdin for reading.
633 For example to read from stdin with ffmpeg:
635 cat test.wav | ffmpeg -i pipe:0
637 # ...this is the same as...
638 cat test.wav | ffmpeg -i pipe:
639 For writing to stdout with ffmpeg:
641 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
643 # ...this is the same as...
644 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
645 This protocol accepts the following options:
647 blocksize
649 Set I/O operation maximum block size, in bytes. Default value is
650 "INT_MAX", which results in not limiting the requested block size.
651 Setting this value reasonably low improves user termination request
652 reaction time, which is valuable if data transmission is slow.
653 Note that some formats (typically MOV), require the output protocol to
655 be seekable, so they will fail with the pipe output protocol.
656 prompeg
658 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
659 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction
661 mechanism for MPEG-2 Transport Streams sent over RTP.
662 This protocol must be used in conjunction with the "rtp_mpegts" muxer
664 and the "rtp" protocol.
665 The required syntax is:
667 -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>
669 The destination UDP ports are "port + 2" for the column FEC stream and
671 "port + 4" for the row FEC stream.
672 This protocol accepts the following options:
674 l=n The number of columns (4-20, LxD <= 100)
676 d=n The number of rows (4-20, LxD <= 100)
678 Example usage:
680 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>
682 rist
684 Reliable Internet Streaming Transport protocol
685 The accepted options are:
687 rist_profile
689 Supported values:
690 simple
692 main
693 This one is default.
694 advanced
696 buffer_size
697 Set internal RIST buffer size in milliseconds for retransmission of
698 data. Default value is 0 which means the librist default (1 sec).
699 Maximum value is 30 seconds.
700 fifo_size
702 Size of the librist receiver output fifo in number of packets. This
703 must be a power of 2. Defaults to 8192 (vs the librist default of
704 1024).
705 overrun_nonfatal=1|0
707 Survive in case of librist fifo buffer overrun. Default value is 0.
708 pkt_size
710 Set maximum packet size for sending data. 1316 by default.
711 log_level
713 Set loglevel for RIST logging messages. You only need to set this
714 if you explicitly want to enable debug level messages or packet
715 loss simulation, otherwise the regular loglevel is respected.
716 secret
718 Set override of encryption secret, by default is unset.
719 encryption
721 Set encryption type, by default is disabled. Acceptable values are
722 128 and 256.
723 rtmp
725 Real-Time Messaging Protocol.
726 The Real-Time Messaging Protocol (RTMP) is used for streaming
728 multimedia content across a TCP/IP network.
729 The required syntax is:
731 rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]
733 The accepted parameters are:
735 username
737 An optional username (mostly for publishing).
738 password
740 An optional password (mostly for publishing).
741 server
743 The address of the RTMP server.
744 port
746 The number of the TCP port to use (by default is 1935).
747 app It is the name of the application to access. It usually corresponds
749 to the path where the application is installed on the RTMP server
750 (e.g. /ondemand/, /flash/live/, etc.). You can override the value
751 parsed from the URI through the "rtmp_app" option, too.
752 playpath
754 It is the path or name of the resource to play with reference to
755 the application specified in app, may be prefixed by "mp4:". You
756 can override the value parsed from the URI through the
757 "rtmp_playpath" option, too.
758 listen
760 Act as a server, listening for an incoming connection.
761 timeout
763 Maximum time to wait for the incoming connection. Implies listen.
764 Additionally, the following parameters can be set via command line
766 options (or in code via "AVOption"s):
767 rtmp_app
769 Name of application to connect on the RTMP server. This option
770 overrides the parameter specified in the URI.
771 rtmp_buffer
773 Set the client buffer time in milliseconds. The default is 3000.
774 rtmp_conn
776 Extra arbitrary AMF connection parameters, parsed from a string,
777 e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0". Each
778 value is prefixed by a single character denoting the type, B for
779 Boolean, N for number, S for string, O for object, or Z for null,
780 followed by a colon. For Booleans the data must be either 0 or 1
781 for FALSE or TRUE, respectively. Likewise for Objects the data
782 must be 0 or 1 to end or begin an object, respectively. Data items
783 in subobjects may be named, by prefixing the type with 'N' and
784 specifying the name before the value (i.e. "NB:myFlag:1"). This
785 option may be used multiple times to construct arbitrary AMF
786 sequences.
787 rtmp_flashver
789 Version of the Flash plugin used to run the SWF player. The default
790 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0
791 (compatible; <libavformat version>).)
792 rtmp_flush_interval
794 Number of packets flushed in the same request (RTMPT only). The
795 default is 10.
796 rtmp_live
798 Specify that the media is a live stream. No resuming or seeking in
799 live streams is possible. The default value is "any", which means
800 the subscriber first tries to play the live stream specified in the
801 playpath. If a live stream of that name is not found, it plays the
802 recorded stream. The other possible values are "live" and
803 "recorded".
804 rtmp_pageurl
806 URL of the web page in which the media was embedded. By default no
807 value will be sent.
808 rtmp_playpath
810 Stream identifier to play or to publish. This option overrides the
811 parameter specified in the URI.
812 rtmp_subscribe
814 Name of live stream to subscribe to. By default no value will be
815 sent. It is only sent if the option is specified or if rtmp_live
816 is set to live.
817 rtmp_swfhash
819 SHA256 hash of the decompressed SWF file (32 bytes).
820 rtmp_swfsize
822 Size of the decompressed SWF file, required for SWFVerification.
823 rtmp_swfurl
825 URL of the SWF player for the media. By default no value will be
826 sent.
827 rtmp_swfverify
829 URL to player swf file, compute hash/size automatically.
830 rtmp_tcurl
832 URL of the target stream. Defaults to proto://host[:port]/app.
833 tcp_nodelay=1|0
835 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
836 Remark: Writing to the socket is currently not optimized to
838 minimize system calls and reduces the efficiency / effect of
839 TCP_NODELAY.
840 For example to read with ffplay a multimedia resource named "sample"
842 from the application "vod" from an RTMP server "myserver":
843 ffplay rtmp://myserver/vod/sample
845 To publish to a password protected server, passing the playpath and app
847 names separately:
848 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/
850 rtmpe
852 Encrypted Real-Time Messaging Protocol.
853 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
855 streaming multimedia content within standard cryptographic primitives,
856 consisting of Diffie-Hellman key exchange and HMACSHA256, generating a
857 pair of RC4 keys.
858 rtmps
860 Real-Time Messaging Protocol over a secure SSL connection.
861 The Real-Time Messaging Protocol (RTMPS) is used for streaming
863 multimedia content across an encrypted connection.
864 rtmpt
866 Real-Time Messaging Protocol tunneled through HTTP.
867 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
869 for streaming multimedia content within HTTP requests to traverse
870 firewalls.
871 rtmpte
873 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
874 The Encrypted Real-Time Messaging Protocol tunneled through HTTP
876 (RTMPTE) is used for streaming multimedia content within HTTP requests
877 to traverse firewalls.
878 rtmpts
880 Real-Time Messaging Protocol tunneled through HTTPS.
881 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is
883 used for streaming multimedia content within HTTPS requests to traverse
884 firewalls.
885 libsmbclient
887 libsmbclient permits one to manipulate CIFS/SMB network resources.
888 Following syntax is required.
890 smb://[[domain:]user[:password@]]server[/share[/path[/file]]]
892 This protocol accepts the following options.
894 timeout
896 Set timeout in milliseconds of socket I/O operations used by the
897 underlying low level operation. By default it is set to -1, which
898 means that the timeout is not specified.
899 truncate
901 Truncate existing files on write, if set to 1. A value of 0
902 prevents truncating. Default value is 1.
903 workgroup
905 Set the workgroup used for making connections. By default workgroup
906 is not specified.
907 For more information see: <http://www.samba.org/>.
909 libssh
911 Secure File Transfer Protocol via libssh
912 Read from or write to remote resources using SFTP protocol.
914 Following syntax is required.
916 sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
918 This protocol accepts the following options.
920 timeout
922 Set timeout of socket I/O operations used by the underlying low
923 level operation. By default it is set to -1, which means that the
924 timeout is not specified.
925 truncate
927 Truncate existing files on write, if set to 1. A value of 0
928 prevents truncating. Default value is 1.
929 private_key
931 Specify the path of the file containing private key to use during
932 authorization. By default libssh searches for keys in the ~/.ssh/
933 directory.
934 Example: Play a file stored on remote server.
936 ffplay sftp://user:password@server_address:22/home/user/resource.mpeg
938 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
940 Real-Time Messaging Protocol and its variants supported through
941 librtmp.
942 Requires the presence of the librtmp headers and library during
944 configuration. You need to explicitly configure the build with
945 "--enable-librtmp". If enabled this will replace the native RTMP
946 protocol.
947 This protocol provides most client functions and a few server functions
949 needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP
950 (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these
951 encrypted types (RTMPTE, RTMPTS).
952 The required syntax is:
954 <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>
956 where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe",
958 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
959 server, port, app and playpath have the same meaning as specified for
960 the RTMP native protocol. options contains a list of space-separated
961 options of the form key=val.
962 See the librtmp manual page (man 3 librtmp) for more information.
964 For example, to stream a file in real-time to an RTMP server using
966 ffmpeg:
967 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
969 To play the same stream using ffplay:
971 ffplay "rtmp://myserver/live/mystream live=1"
973 rtp
975 Real-time Transport Protocol.
976 The required syntax for an RTP URL is:
978 rtp://hostname[:port][?option=val...]
979 port specifies the RTP port to use.
981 The following URL options are supported:
983 ttl=n
985 Set the TTL (Time-To-Live) value (for multicast only).
986 rtcpport=n
988 Set the remote RTCP port to n.
989 localrtpport=n
991 Set the local RTP port to n.
992 localrtcpport=n'
994 Set the local RTCP port to n.
995 pkt_size=n
997 Set max packet size (in bytes) to n.
998 buffer_size=size
1000 Set the maximum UDP socket buffer size in bytes.
1001 connect=0|1
1003 Do a "connect()" on the UDP socket (if set to 1) or not (if set to
1004 0).
1005 sources=ip[,ip]
1007 List allowed source IP addresses.
1008 block=ip[,ip]
1010 List disallowed (blocked) source IP addresses.
1011 write_to_source=0|1
1013 Send packets to the source address of the latest received packet
1014 (if set to 1) or to a default remote address (if set to 0).
1015 localport=n
1017 Set the local RTP port to n.
1018 localaddr=addr
1020 Local IP address of a network interface used for sending packets or
1021 joining multicast groups.
1022 timeout=n
1024 Set timeout (in microseconds) of socket I/O operations to n.
1025 This is a deprecated option. Instead, localrtpport should be used.
1027 Important notes:
1029 1. If rtcpport is not set the RTCP port will be set to the RTP port
1031 value plus 1.
1032 2. If localrtpport (the local RTP port) is not set any available port
1034 will be used for the local RTP and RTCP ports.
1035 3. If localrtcpport (the local RTCP port) is not set it will be set to
1037 the local RTP port value plus 1.
1038 rtsp
1040 Real-Time Streaming Protocol.
1041 RTSP is not technically a protocol handler in libavformat, it is a
1043 demuxer and muxer. The demuxer supports both normal RTSP (with data
1044 transferred over RTP; this is used by e.g. Apple and Microsoft) and
1045 Real-RTSP (with data transferred over RDT).
1046 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
1048 supporting it (currently Darwin Streaming Server and Mischa
1049 Spiegelmock's <https://github.com/revmischa/rtsp-server>).
1050 The required syntax for a RTSP url is:
1052 rtsp://<hostname>[:<port>]/<path>
1054 Options can be set on the ffmpeg/ffplay command line, or set in code
1056 via "AVOption"s or in "avformat_open_input".
1057 The following options are supported.
1059 initial_pause
1061 Do not start playing the stream immediately if set to 1. Default
1062 value is 0.
1063 rtsp_transport
1065 Set RTSP transport protocols.
1066 It accepts the following values:
1068 udp Use UDP as lower transport protocol.
1070 tcp Use TCP (interleaving within the RTSP control channel) as lower
1072 transport protocol.
1073 udp_multicast
1075 Use UDP multicast as lower transport protocol.
1076 http
1078 Use HTTP tunneling as lower transport protocol, which is useful
1079 for passing proxies.
1080 Multiple lower transport protocols may be specified, in that case
1082 they are tried one at a time (if the setup of one fails, the next
1083 one is tried). For the muxer, only the tcp and udp options are
1084 supported.
1085 rtsp_flags
1087 Set RTSP flags.
1088 The following values are accepted:
1090 filter_src
1092 Accept packets only from negotiated peer address and port.
1093 listen
1095 Act as a server, listening for an incoming connection.
1096 prefer_tcp
1098 Try TCP for RTP transport first, if TCP is available as RTSP
1099 RTP transport.
1100 Default value is none.
1102 allowed_media_types
1104 Set media types to accept from the server.
1105 The following flags are accepted:
1107 video
1109 audio
1110 data
1111 By default it accepts all media types.
1113 min_port
1115 Set minimum local UDP port. Default value is 5000.
1116 max_port
1118 Set maximum local UDP port. Default value is 65000.
1119 listen_timeout
1121 Set maximum timeout (in seconds) to establish an initial
1122 connection. Setting listen_timeout > 0 sets rtsp_flags to listen.
1123 Default is -1 which means an infinite timeout when listen mode is
1124 set.
1125 reorder_queue_size
1127 Set number of packets to buffer for handling of reordered packets.
1128 timeout
1130 Set socket TCP I/O timeout in microseconds.
1131 user_agent
1133 Override User-Agent header. If not specified, it defaults to the
1134 libavformat identifier string.
1135 When receiving data over UDP, the demuxer tries to reorder received
1137 packets (since they may arrive out of order, or packets may get lost
1138 totally). This can be disabled by setting the maximum demuxing delay to
1139 zero (via the "max_delay" field of AVFormatContext).
1140 When watching multi-bitrate Real-RTSP streams with ffplay, the streams
1142 to display can be chosen with "-vst" n and "-ast" n for video and audio
1143 respectively, and can be switched on the fly by pressing "v" and "a".
1144 Examples
1146 The following examples all make use of the ffplay and ffmpeg tools.
1148 • Watch a stream over UDP, with a max reordering delay of 0.5
1150 seconds:
1151 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1153 • Watch a stream tunneled over HTTP:
1155 ffplay -rtsp_transport http rtsp://server/video.mp4
1157 • Send a stream in realtime to a RTSP server, for others to watch:
1159 ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1161 • Receive a stream in realtime:
1163 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>
1165 sap
1167 Session Announcement Protocol (RFC 2974). This is not technically a
1168 protocol handler in libavformat, it is a muxer and demuxer. It is used
1169 for signalling of RTP streams, by announcing the SDP for the streams
1170 regularly on a separate port.
1171 Muxer
1173 The syntax for a SAP url given to the muxer is:
1175 sap://<destination>[:<port>][?<options>]
1177 The RTP packets are sent to destination on port port, or to port 5004
1179 if no port is specified. options is a "&"-separated list. The
1180 following options are supported:
1181 announce_addr=address
1183 Specify the destination IP address for sending the announcements
1184 to. If omitted, the announcements are sent to the commonly used
1185 SAP announcement multicast address 224.2.127.254 (sap.mcast.net),
1186 or ff0e::2:7ffe if destination is an IPv6 address.
1187 announce_port=port
1189 Specify the port to send the announcements on, defaults to 9875 if
1190 not specified.
1191 ttl=ttl
1193 Specify the time to live value for the announcements and RTP
1194 packets, defaults to 255.
1195 same_port=0|1
1197 If set to 1, send all RTP streams on the same port pair. If zero
1198 (the default), all streams are sent on unique ports, with each
1199 stream on a port 2 numbers higher than the previous. VLC/Live555
1200 requires this to be set to 1, to be able to receive the stream.
1201 The RTP stack in libavformat for receiving requires all streams to
1202 be sent on unique ports.
1203 Example command lines follow.
1205 To broadcast a stream on the local subnet, for watching in VLC:
1207 ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1
1209 Similarly, for watching in ffplay:
1211 ffmpeg -re -i <input> -f sap sap://224.0.0.255
1213 And for watching in ffplay, over IPv6:
1215 ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]
1217 Demuxer
1219 The syntax for a SAP url given to the demuxer is:
1221 sap://[<address>][:<port>]
1223 address is the multicast address to listen for announcements on, if
1225 omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the
1226 port that is listened on, 9875 if omitted.
1227 The demuxers listens for announcements on the given address and port.
1229 Once an announcement is received, it tries to receive that particular
1230 stream.
1231 Example command lines follow.
1233 To play back the first stream announced on the normal SAP multicast
1235 address:
1236 ffplay sap://
1238 To play back the first stream announced on one the default IPv6 SAP
1240 multicast address:
1241 ffplay sap://[ff0e::2:7ffe]
1243 sctp
1245 Stream Control Transmission Protocol.
1246 The accepted URL syntax is:
1248 sctp://<host>:<port>[?<options>]
1250 The protocol accepts the following options:
1252 listen
1254 If set to any value, listen for an incoming connection. Outgoing
1255 connection is done by default.
1256 max_streams
1258 Set the maximum number of streams. By default no limit is set.
1259 srt
1261 Haivision Secure Reliable Transport Protocol via libsrt.
1262 The supported syntax for a SRT URL is:
1264 srt://<hostname>:<port>[?<options>]
1266 options contains a list of &-separated options of the form key=val.
1268 or
1270 <options> srt://<hostname>:<port>
1272 options contains a list of '-key val' options.
1274 This protocol accepts the following options.
1276 connect_timeout=milliseconds
1278 Connection timeout; SRT cannot connect for RTT > 1500 msec (2
1279 handshake exchanges) with the default connect timeout of 3 seconds.
1280 This option applies to the caller and rendezvous connection modes.
1281 The connect timeout is 10 times the value set for the rendezvous
1282 mode (which can be used as a workaround for this connection problem
1283 with earlier versions).
1284 ffs=bytes
1286 Flight Flag Size (Window Size), in bytes. FFS is actually an
1287 internal parameter and you should set it to not less than
1288 recv_buffer_size and mss. The default value is relatively large,
1289 therefore unless you set a very large receiver buffer, you do not
1290 need to change this option. Default value is 25600.
1291 inputbw=bytes/seconds
1293 Sender nominal input rate, in bytes per seconds. Used along with
1294 oheadbw, when maxbw is set to relative (0), to calculate maximum
1295 sending rate when recovery packets are sent along with the main
1296 media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set
1297 while maxbw is set to relative (0), the actual input rate is
1298 evaluated inside the library. Default value is 0.
1299 iptos=tos
1301 IP Type of Service. Applies to sender only. Default value is 0xB8.
1302 ipttl=ttl
1304 IP Time To Live. Applies to sender only. Default value is 64.
1305 latency=microseconds
1307 Timestamp-based Packet Delivery Delay. Used to absorb bursts of
1308 missed packet retransmissions. This flag sets both rcvlatency and
1309 peerlatency to the same value. Note that prior to version 1.3.0
1310 this is the only flag to set the latency, however this is
1311 effectively equivalent to setting peerlatency, when side is sender
1312 and rcvlatency when side is receiver, and the bidirectional stream
1313 sending is not supported.
1314 listen_timeout=microseconds
1316 Set socket listen timeout.
1317 maxbw=bytes/seconds
1319 Maximum sending bandwidth, in bytes per seconds. -1 infinite
1320 (CSRTCC limit is 30mbps) 0 relative to input rate (see inputbw) >0
1321 absolute limit value Default value is 0 (relative)
1322 mode=caller|listener|rendezvous
1324 Connection mode. caller opens client connection. listener starts
1325 server to listen for incoming connections. rendezvous use Rendez-
1326 Vous connection mode. Default value is caller.
1327 mss=bytes
1329 Maximum Segment Size, in bytes. Used for buffer allocation and rate
1330 calculation using a packet counter assuming fully filled packets.
1331 The smallest MSS between the peers is used. This is 1500 by default
1332 in the overall internet. This is the maximum size of the UDP
1333 packet and can be only decreased, unless you have some unusual
1334 dedicated network settings. Default value is 1500.
1335 nakreport=1|0
1337 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1338 periodically until a lost packet is retransmitted or intentionally
1339 dropped. Default value is 1.
1340 oheadbw=percents
1342 Recovery bandwidth overhead above input rate, in percents. See
1343 inputbw. Default value is 25%.
1344 passphrase=string
1346 HaiCrypt Encryption/Decryption Passphrase string, length from 10 to
1347 79 characters. The passphrase is the shared secret between the
1348 sender and the receiver. It is used to generate the Key Encrypting
1349 Key using PBKDF2 (Password-Based Key Derivation Function). It is
1350 used only if pbkeylen is non-zero. It is used on the receiver only
1351 if the received data is encrypted. The configured passphrase
1352 cannot be recovered (write-only).
1353 enforced_encryption=1|0
1355 If true, both connection parties must have the same password set
1356 (including empty, that is, with no encryption). If the password
1357 doesn't match or only one side is unencrypted, the connection is
1358 rejected. Default is true.
1359 kmrefreshrate=packets
1361 The number of packets to be transmitted after which the encryption
1362 key is switched to a new key. Default is -1. -1 means auto
1363 (0x1000000 in srt library). The range for this option is integers
1364 in the 0 - "INT_MAX".
1365 kmpreannounce=packets
1367 The interval between when a new encryption key is sent and when
1368 switchover occurs. This value also applies to the subsequent
1369 interval between when switchover occurs and when the old encryption
1370 key is decommissioned. Default is -1. -1 means auto (0x1000 in srt
1371 library). The range for this option is integers in the 0 -
1372 "INT_MAX".
1373 snddropdelay=microseconds
1375 The sender's extra delay before dropping packets. This delay is
1376 added to the default drop delay time interval value.
1377 Special value -1: Do not drop packets on the sender at all.
1379 payload_size=bytes
1381 Sets the maximum declared size of a packet transferred during the
1382 single call to the sending function in Live mode. Use 0 if this
1383 value isn't used (which is default in file mode). Default is -1
1384 (automatic), which typically means MPEG-TS; if you are going to use
1385 SRT to send any different kind of payload, such as, for example,
1386 wrapping a live stream in very small frames, then you can use a
1387 bigger maximum frame size, though not greater than 1456 bytes.
1388 pkt_size=bytes
1390 Alias for payload_size.
1391 peerlatency=microseconds
1393 The latency value (as described in rcvlatency) that is set by the
1394 sender side as a minimum value for the receiver.
1395 pbkeylen=bytes
1397 Sender encryption key length, in bytes. Only can be set to 0, 16,
1398 24 and 32. Enable sender encryption if not 0. Not required on
1399 receiver (set to 0), key size obtained from sender in HaiCrypt
1400 handshake. Default value is 0.
1401 rcvlatency=microseconds
1403 The time that should elapse since the moment when the packet was
1404 sent and the moment when it's delivered to the receiver application
1405 in the receiving function. This time should be a buffer time large
1406 enough to cover the time spent for sending, unexpectedly extended
1407 RTT time, and the time needed to retransmit the lost UDP packet.
1408 The effective latency value will be the maximum of this options'
1409 value and the value of peerlatency set by the peer side. Before
1410 version 1.3.0 this option is only available as latency.
1411 recv_buffer_size=bytes
1413 Set UDP receive buffer size, expressed in bytes.
1414 send_buffer_size=bytes
1416 Set UDP send buffer size, expressed in bytes.
1417 timeout=microseconds
1419 Set raise error timeouts for read, write and connect operations.
1420 Note that the SRT library has internal timeouts which can be
1421 controlled separately, the value set here is only a cap on those.
1422 tlpktdrop=1|0
1424 Too-late Packet Drop. When enabled on receiver, it skips missing
1425 packets that have not been delivered in time and delivers the
1426 following packets to the application when their time-to-play has
1427 come. It also sends a fake ACK to the sender. When enabled on
1428 sender and enabled on the receiving peer, the sender drops the
1429 older packets that have no chance of being delivered in time. It
1430 was automatically enabled in the sender if the receiver supports
1431 it.
1432 sndbuf=bytes
1434 Set send buffer size, expressed in bytes.
1435 rcvbuf=bytes
1437 Set receive buffer size, expressed in bytes.
1438 Receive buffer must not be greater than ffs.
1440 lossmaxttl=packets
1442 The value up to which the Reorder Tolerance may grow. When Reorder
1443 Tolerance is > 0, then packet loss report is delayed until that
1444 number of packets come in. Reorder Tolerance increases every time a
1445 "belated" packet has come, but it wasn't due to retransmission
1446 (that is, when UDP packets tend to come out of order), with the
1447 difference between the latest sequence and this packet's sequence,
1448 and not more than the value of this option. By default it's 0,
1449 which means that this mechanism is turned off, and the loss report
1450 is always sent immediately upon experiencing a "gap" in sequences.
1451 minversion
1453 The minimum SRT version that is required from the peer. A
1454 connection to a peer that does not satisfy the minimum version
1455 requirement will be rejected.
1456 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1458 form.
1459 streamid=string
1461 A string limited to 512 characters that can be set on the socket
1462 prior to connecting. This stream ID will be able to be retrieved by
1463 the listener side from the socket that is returned from srt_accept
1464 and was connected by a socket with that set stream ID. SRT does not
1465 enforce any special interpretation of the contents of this string.
1466 This option doesnXt make sense in Rendezvous connection; the result
1467 might be that simply one side will override the value from the
1468 other side and itXs the matter of luck which one would win
1469 srt_streamid=string
1471 Alias for streamid to avoid conflict with ffmpeg command line
1472 option.
1473 smoother=live|file
1475 The type of Smoother used for the transmission for that socket,
1476 which is responsible for the transmission and congestion control.
1477 The Smoother type must be exactly the same on both connecting
1478 parties, otherwise the connection is rejected.
1479 messageapi=1|0
1481 When set, this socket uses the Message API, otherwise it uses
1482 Buffer API. Note that in live mode (see transtype) thereXs only
1483 message API available. In File mode you can chose to use one of two
1484 modes:
1485 Stream API (default, when this option is false). In this mode you
1487 may send as many data as you wish with one sending instruction, or
1488 even use dedicated functions that read directly from a file. The
1489 internal facility will take care of any speed and congestion
1490 control. When receiving, you can also receive as many data as
1491 desired, the data not extracted will be waiting for the next call.
1492 There is no boundary between data portions in the Stream mode.
1493 Message API. In this mode your single sending instruction passes
1495 exactly one piece of data that has boundaries (a message). Contrary
1496 to Live mode, this message may span across multiple UDP packets and
1497 the only size limitation is that it shall fit as a whole in the
1498 sending buffer. The receiver shall use as large buffer as necessary
1499 to receive the message, otherwise the message will not be given up.
1500 When the message is not complete (not all packets received or there
1501 was a packet loss) it will not be given up.
1502 transtype=live|file
1504 Sets the transmission type for the socket, in particular, setting
1505 this option sets multiple other parameters to their default values
1506 as required for a particular transmission type.
1507 live: Set options as for live transmission. In this mode, you
1509 should send by one sending instruction only so many data that fit
1510 in one UDP packet, and limited to the value defined first in
1511 payload_size (1316 is default in this mode). There is no speed
1512 control in this mode, only the bandwidth control, if configured, in
1513 order to not exceed the bandwidth with the overhead transmission
1514 (retransmitted and control packets).
1515 file: Set options as for non-live transmission. See messageapi for
1517 further explanations
1518 linger=seconds
1520 The number of seconds that the socket waits for unsent data when
1521 closing. Default is -1. -1 means auto (off with 0 seconds in live
1522 mode, on with 180 seconds in file mode). The range for this option
1523 is integers in the 0 - "INT_MAX".
1524 tsbpd=1|0
1526 When true, use Timestamp-based Packet Delivery mode. The default
1527 behavior depends on the transmission type: enabled in live mode,
1528 disabled in file mode.
1529 For more information see: <https://github.com/Haivision/srt>.
1531 srtp
1533 Secure Real-time Transport Protocol.
1534 The accepted options are:
1536 srtp_in_suite
1538 srtp_out_suite
1539 Select input and output encoding suites.
1540 Supported values:
1542 AES_CM_128_HMAC_SHA1_80
1544 SRTP_AES128_CM_HMAC_SHA1_80
1545 AES_CM_128_HMAC_SHA1_32
1546 SRTP_AES128_CM_HMAC_SHA1_32
1547 srtp_in_params
1548 srtp_out_params
1549 Set input and output encoding parameters, which are expressed by a
1550 base64-encoded representation of a binary block. The first 16 bytes
1551 of this binary block are used as master key, the following 14 bytes
1552 are used as master salt.
1553 subfile
1555 Virtually extract a segment of a file or another stream. The
1556 underlying stream must be seekable.
1557 Accepted options:
1559 start
1561 Start offset of the extracted segment, in bytes.
1562 end End offset of the extracted segment, in bytes. If set to 0,
1564 extract till end of file.
1565 Examples:
1567 Extract a chapter from a DVD VOB file (start and end sectors obtained
1569 externally and multiplied by 2048):
1570 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1572 Play an AVI file directly from a TAR archive:
1574 subfile,,start,183241728,end,366490624,,:archive.tar
1576 Play a MPEG-TS file from start offset till end:
1578 subfile,,start,32815239,end,0,,:video.ts
1580 tee
1582 Writes the output to multiple protocols. The individual outputs are
1583 separated by |
1584 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1586 tcp
1588 Transmission Control Protocol.
1589 The required syntax for a TCP url is:
1591 tcp://<hostname>:<port>[?<options>]
1593 options contains a list of &-separated options of the form key=val.
1595 The list of supported options follows.
1597 listen=2|1|0
1599 Listen for an incoming connection. 0 disables listen, 1 enables
1600 listen in single client mode, 2 enables listen in multi-client
1601 mode. Default value is 0.
1602 timeout=microseconds
1604 Set raise error timeout, expressed in microseconds.
1605 This option is only relevant in read mode: if no data arrived in
1607 more than this time interval, raise error.
1608 listen_timeout=milliseconds
1610 Set listen timeout, expressed in milliseconds.
1611 recv_buffer_size=bytes
1613 Set receive buffer size, expressed bytes.
1614 send_buffer_size=bytes
1616 Set send buffer size, expressed bytes.
1617 tcp_nodelay=1|0
1619 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1620 Remark: Writing to the socket is currently not optimized to
1622 minimize system calls and reduces the efficiency / effect of
1623 TCP_NODELAY.
1624 tcp_mss=bytes
1626 Set maximum segment size for outgoing TCP packets, expressed in
1627 bytes.
1628 The following example shows how to setup a listening TCP connection
1630 with ffmpeg, which is then accessed with ffplay:
1631 ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
1633 ffplay tcp://<hostname>:<port>
1634 tls
1636 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1637 The required syntax for a TLS/SSL url is:
1639 tls://<hostname>:<port>[?<options>]
1641 The following parameters can be set via command line options (or in
1643 code via "AVOption"s):
1644 ca_file, cafile=filename
1646 A file containing certificate authority (CA) root certificates to
1647 treat as trusted. If the linked TLS library contains a default this
1648 might not need to be specified for verification to work, but not
1649 all libraries and setups have defaults built in. The file must be
1650 in OpenSSL PEM format.
1651 tls_verify=1|0
1653 If enabled, try to verify the peer that we are communicating with.
1654 Note, if using OpenSSL, this currently only makes sure that the
1655 peer certificate is signed by one of the root certificates in the
1656 CA database, but it does not validate that the certificate actually
1657 matches the host name we are trying to connect to. (With other
1658 backends, the host name is validated as well.)
1659 This is disabled by default since it requires a CA database to be
1661 provided by the caller in many cases.
1662 cert_file, cert=filename
1664 A file containing a certificate to use in the handshake with the
1665 peer. (When operating as server, in listen mode, this is more
1666 often required by the peer, while client certificates only are
1667 mandated in certain setups.)
1668 key_file, key=filename
1670 A file containing the private key for the certificate.
1671 listen=1|0
1673 If enabled, listen for connections on the provided port, and assume
1674 the server role in the handshake instead of the client role.
1675 http_proxy
1677 The HTTP proxy to tunnel through, e.g. "http://example.com:1234".
1678 The proxy must support the CONNECT method.
1679 Example command lines:
1681 To create a TLS/SSL server that serves an input stream.
1683 ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>
1685 To play back a stream from the TLS/SSL server using ffplay:
1687 ffplay tls://<hostname>:<port>
1689 udp
1691 User Datagram Protocol.
1692 The required syntax for an UDP URL is:
1694 udp://<hostname>:<port>[?<options>]
1696 options contains a list of &-separated options of the form key=val.
1698 In case threading is enabled on the system, a circular buffer is used
1700 to store the incoming data, which allows one to reduce loss of data due
1701 to UDP socket buffer overruns. The fifo_size and overrun_nonfatal
1702 options are related to this buffer.
1703 The list of supported options follows.
1705 buffer_size=size
1707 Set the UDP maximum socket buffer size in bytes. This is used to
1708 set either the receive or send buffer size, depending on what the
1709 socket is used for. Default is 32 KB for output, 384 KB for input.
1710 See also fifo_size.
1711 bitrate=bitrate
1713 If set to nonzero, the output will have the specified constant
1714 bitrate if the input has enough packets to sustain it.
1715 burst_bits=bits
1717 When using bitrate this specifies the maximum number of bits in
1718 packet bursts.
1719 localport=port
1721 Override the local UDP port to bind with.
1722 localaddr=addr
1724 Local IP address of a network interface used for sending packets or
1725 joining multicast groups.
1726 pkt_size=size
1728 Set the size in bytes of UDP packets.
1729 reuse=1|0
1731 Explicitly allow or disallow reusing UDP sockets.
1732 ttl=ttl
1734 Set the time to live value (for multicast only).
1735 connect=1|0
1737 Initialize the UDP socket with "connect()". In this case, the
1738 destination address can't be changed with ff_udp_set_remote_url
1739 later. If the destination address isn't known at the start, this
1740 option can be specified in ff_udp_set_remote_url, too. This allows
1741 finding out the source address for the packets with getsockname,
1742 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1743 unreachable" is received. For receiving, this gives the benefit of
1744 only receiving packets from the specified peer address/port.
1745 sources=address[,address]
1747 Only receive packets sent from the specified addresses. In case of
1748 multicast, also subscribe to multicast traffic coming from these
1749 addresses only.
1750 block=address[,address]
1752 Ignore packets sent from the specified addresses. In case of
1753 multicast, also exclude the source addresses in the multicast
1754 subscription.
1755 fifo_size=units
1757 Set the UDP receiving circular buffer size, expressed as a number
1758 of packets with size of 188 bytes. If not specified defaults to
1759 7*4096.
1760 overrun_nonfatal=1|0
1762 Survive in case of UDP receiving circular buffer overrun. Default
1763 value is 0.
1764 timeout=microseconds
1766 Set raise error timeout, expressed in microseconds.
1767 This option is only relevant in read mode: if no data arrived in
1769 more than this time interval, raise error.
1770 broadcast=1|0
1772 Explicitly allow or disallow UDP broadcasting.
1773 Note that broadcasting may not work properly on networks having a
1775 broadcast storm protection.
1776 Examples
1778 • Use ffmpeg to stream over UDP to a remote endpoint:
1780 ffmpeg -i <input> -f <format> udp://<hostname>:<port>
1782 • Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP
1784 packets, using a large input buffer:
1785 ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535
1787 • Use ffmpeg to receive over UDP from a remote endpoint:
1789 ffmpeg -i udp://[<multicast-address>]:<port> ...
1791 unix
1793 Unix local socket
1794 The required syntax for a Unix socket URL is:
1796 unix://<filepath>
1798 The following parameters can be set via command line options (or in
1800 code via "AVOption"s):
1801 timeout
1803 Timeout in ms.
1804 listen
1806 Create the Unix socket in listening mode.
1807 zmq
1809 ZeroMQ asynchronous messaging using the libzmq library.
1810 This library supports unicast streaming to multiple clients without
1812 relying on an external server.
1813 The required syntax for streaming or connecting to a stream is:
1815 zmq:tcp://ip-address:port
1817 Example: Create a localhost stream on port 5555:
1819 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1821 Multiple clients may connect to the stream using:
1823 ffplay zmq:tcp://127.0.0.1:5555
1825 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub
1827 pattern. The server side binds to a port and publishes data. Clients
1828 connect to the server (via IP address/port) and subscribe to the
1829 stream. The order in which the server and client start generally does
1830 not matter.
1831 ffmpeg must be compiled with the --enable-libzmq option to support this
1833 protocol.
1834 Options can be set on the ffmpeg/ffplay command line. The following
1836 options are supported:
1837 pkt_size
1839 Forces the maximum packet size for sending/receiving data. The
1840 default value is 131,072 bytes. On the server side, this sets the
1841 maximum size of sent packets via ZeroMQ. On the clients, it sets an
1842 internal buffer size for receiving packets. Note that pkt_size on
1843 the clients should be equal to or greater than pkt_size on the
1844 server. Otherwise the received message may be truncated causing
1845 decoding errors.
1846
1848 ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
1849
1851 The FFmpeg developers.
1852 For details about the authorship, see the Git history of the project
1854 (git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
1855 the FFmpeg source directory, or browsing the online repository at
1856 <http://source.ffmpeg.org>.
1857 Maintainers for the specific components are listed in the file
1859 MAINTAINERS in the source code tree.
1860 FFMPEG-PROTOCOLS(1)