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 mmst
566 MMS (Microsoft Media Server) protocol over TCP.
567 mmsh
569 MMS (Microsoft Media Server) protocol over HTTP.
570 The required syntax is:
572 mmsh://<server>[:<port>][/<app>][/<playpath>]
574 md5
576 MD5 output protocol.
577 Computes the MD5 hash of the data to be written, and on close writes
579 this to the designated output or stdout if none is specified. It can be
580 used to test muxers without writing an actual file.
581 Some examples follow.
583 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
585 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
586 # Write the MD5 hash of the encoded AVI file to stdout.
588 ffmpeg -i input.flv -f avi -y md5:
589 Note that some formats (typically MOV) require the output protocol to
591 be seekable, so they will fail with the MD5 output protocol.
592 pipe
594 UNIX pipe access protocol.
595 Read and write from UNIX pipes.
597 The accepted syntax is:
599 pipe:[<number>]
601 number is the number corresponding to the file descriptor of the pipe
603 (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not
604 specified, by default the stdout file descriptor will be used for
605 writing, stdin for reading.
606 For example to read from stdin with ffmpeg:
608 cat test.wav | ffmpeg -i pipe:0
610 # ...this is the same as...
611 cat test.wav | ffmpeg -i pipe:
612 For writing to stdout with ffmpeg:
614 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
616 # ...this is the same as...
617 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
618 This protocol accepts the following options:
620 blocksize
622 Set I/O operation maximum block size, in bytes. Default value is
623 "INT_MAX", which results in not limiting the requested block size.
624 Setting this value reasonably low improves user termination request
625 reaction time, which is valuable if data transmission is slow.
626 Note that some formats (typically MOV), require the output protocol to
628 be seekable, so they will fail with the pipe output protocol.
629 prompeg
631 Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
632 The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction
634 mechanism for MPEG-2 Transport Streams sent over RTP.
635 This protocol must be used in conjunction with the "rtp_mpegts" muxer
637 and the "rtp" protocol.
638 The required syntax is:
640 -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>
642 The destination UDP ports are "port + 2" for the column FEC stream and
644 "port + 4" for the row FEC stream.
645 This protocol accepts the following options:
647 l=n The number of columns (4-20, LxD <= 100)
649 d=n The number of rows (4-20, LxD <= 100)
651 Example usage:
653 -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>
655 rist
657 Reliable Internet Streaming Transport protocol
658 The accepted options are:
660 rist_profile
662 Supported values:
663 simple
665 main
666 This one is default.
667 advanced
669 buffer_size
670 Set internal RIST buffer size in milliseconds for retransmission of
671 data. Default value is 0 which means the librist default (1 sec).
672 Maximum value is 30 seconds.
673 pkt_size
675 Set maximum packet size for sending data. 1316 by default.
676 log_level
678 Set loglevel for RIST logging messages. You only need to set this
679 if you explicitly want to enable debug level messages or packet
680 loss simulation, otherwise the regular loglevel is respected.
681 secret
683 Set override of encryption secret, by default is unset.
684 encryption
686 Set encryption type, by default is disabled. Acceptable values are
687 128 and 256.
688 rtmp
690 Real-Time Messaging Protocol.
691 The Real-Time Messaging Protocol (RTMP) is used for streaming
693 multimedia content across a TCP/IP network.
694 The required syntax is:
696 rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]
698 The accepted parameters are:
700 username
702 An optional username (mostly for publishing).
703 password
705 An optional password (mostly for publishing).
706 server
708 The address of the RTMP server.
709 port
711 The number of the TCP port to use (by default is 1935).
712 app It is the name of the application to access. It usually corresponds
714 to the path where the application is installed on the RTMP server
715 (e.g. /ondemand/, /flash/live/, etc.). You can override the value
716 parsed from the URI through the "rtmp_app" option, too.
717 playpath
719 It is the path or name of the resource to play with reference to
720 the application specified in app, may be prefixed by "mp4:". You
721 can override the value parsed from the URI through the
722 "rtmp_playpath" option, too.
723 listen
725 Act as a server, listening for an incoming connection.
726 timeout
728 Maximum time to wait for the incoming connection. Implies listen.
729 Additionally, the following parameters can be set via command line
731 options (or in code via "AVOption"s):
732 rtmp_app
734 Name of application to connect on the RTMP server. This option
735 overrides the parameter specified in the URI.
736 rtmp_buffer
738 Set the client buffer time in milliseconds. The default is 3000.
739 rtmp_conn
741 Extra arbitrary AMF connection parameters, parsed from a string,
742 e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0". Each
743 value is prefixed by a single character denoting the type, B for
744 Boolean, N for number, S for string, O for object, or Z for null,
745 followed by a colon. For Booleans the data must be either 0 or 1
746 for FALSE or TRUE, respectively. Likewise for Objects the data
747 must be 0 or 1 to end or begin an object, respectively. Data items
748 in subobjects may be named, by prefixing the type with 'N' and
749 specifying the name before the value (i.e. "NB:myFlag:1"). This
750 option may be used multiple times to construct arbitrary AMF
751 sequences.
752 rtmp_flashver
754 Version of the Flash plugin used to run the SWF player. The default
755 is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0
756 (compatible; <libavformat version>).)
757 rtmp_flush_interval
759 Number of packets flushed in the same request (RTMPT only). The
760 default is 10.
761 rtmp_live
763 Specify that the media is a live stream. No resuming or seeking in
764 live streams is possible. The default value is "any", which means
765 the subscriber first tries to play the live stream specified in the
766 playpath. If a live stream of that name is not found, it plays the
767 recorded stream. The other possible values are "live" and
768 "recorded".
769 rtmp_pageurl
771 URL of the web page in which the media was embedded. By default no
772 value will be sent.
773 rtmp_playpath
775 Stream identifier to play or to publish. This option overrides the
776 parameter specified in the URI.
777 rtmp_subscribe
779 Name of live stream to subscribe to. By default no value will be
780 sent. It is only sent if the option is specified or if rtmp_live
781 is set to live.
782 rtmp_swfhash
784 SHA256 hash of the decompressed SWF file (32 bytes).
785 rtmp_swfsize
787 Size of the decompressed SWF file, required for SWFVerification.
788 rtmp_swfurl
790 URL of the SWF player for the media. By default no value will be
791 sent.
792 rtmp_swfverify
794 URL to player swf file, compute hash/size automatically.
795 rtmp_tcurl
797 URL of the target stream. Defaults to proto://host[:port]/app.
798 tcp_nodelay=1|0
800 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
801 Remark: Writing to the socket is currently not optimized to
803 minimize system calls and reduces the efficiency / effect of
804 TCP_NODELAY.
805 For example to read with ffplay a multimedia resource named "sample"
807 from the application "vod" from an RTMP server "myserver":
808 ffplay rtmp://myserver/vod/sample
810 To publish to a password protected server, passing the playpath and app
812 names separately:
813 ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/
815 rtmpe
817 Encrypted Real-Time Messaging Protocol.
818 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
820 streaming multimedia content within standard cryptographic primitives,
821 consisting of Diffie-Hellman key exchange and HMACSHA256, generating a
822 pair of RC4 keys.
823 rtmps
825 Real-Time Messaging Protocol over a secure SSL connection.
826 The Real-Time Messaging Protocol (RTMPS) is used for streaming
828 multimedia content across an encrypted connection.
829 rtmpt
831 Real-Time Messaging Protocol tunneled through HTTP.
832 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
834 for streaming multimedia content within HTTP requests to traverse
835 firewalls.
836 rtmpte
838 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
839 The Encrypted Real-Time Messaging Protocol tunneled through HTTP
841 (RTMPTE) is used for streaming multimedia content within HTTP requests
842 to traverse firewalls.
843 rtmpts
845 Real-Time Messaging Protocol tunneled through HTTPS.
846 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is
848 used for streaming multimedia content within HTTPS requests to traverse
849 firewalls.
850 libsmbclient
852 libsmbclient permits one to manipulate CIFS/SMB network resources.
853 Following syntax is required.
855 smb://[[domain:]user[:password@]]server[/share[/path[/file]]]
857 This protocol accepts the following options.
859 timeout
861 Set timeout in milliseconds of socket I/O operations used by the
862 underlying low level operation. By default it is set to -1, which
863 means that the timeout is not specified.
864 truncate
866 Truncate existing files on write, if set to 1. A value of 0
867 prevents truncating. Default value is 1.
868 workgroup
870 Set the workgroup used for making connections. By default workgroup
871 is not specified.
872 For more information see: <http://www.samba.org/>.
874 libssh
876 Secure File Transfer Protocol via libssh
877 Read from or write to remote resources using SFTP protocol.
879 Following syntax is required.
881 sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
883 This protocol accepts the following options.
885 timeout
887 Set timeout of socket I/O operations used by the underlying low
888 level operation. By default it is set to -1, which means that the
889 timeout is not specified.
890 truncate
892 Truncate existing files on write, if set to 1. A value of 0
893 prevents truncating. Default value is 1.
894 private_key
896 Specify the path of the file containing private key to use during
897 authorization. By default libssh searches for keys in the ~/.ssh/
898 directory.
899 Example: Play a file stored on remote server.
901 ffplay sftp://user:password@server_address:22/home/user/resource.mpeg
903 librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
905 Real-Time Messaging Protocol and its variants supported through
906 librtmp.
907 Requires the presence of the librtmp headers and library during
909 configuration. You need to explicitly configure the build with
910 "--enable-librtmp". If enabled this will replace the native RTMP
911 protocol.
912 This protocol provides most client functions and a few server functions
914 needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP
915 (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these
916 encrypted types (RTMPTE, RTMPTS).
917 The required syntax is:
919 <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>
921 where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe",
923 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
924 server, port, app and playpath have the same meaning as specified for
925 the RTMP native protocol. options contains a list of space-separated
926 options of the form key=val.
927 See the librtmp manual page (man 3 librtmp) for more information.
929 For example, to stream a file in real-time to an RTMP server using
931 ffmpeg:
932 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
934 To play the same stream using ffplay:
936 ffplay "rtmp://myserver/live/mystream live=1"
938 rtp
940 Real-time Transport Protocol.
941 The required syntax for an RTP URL is:
943 rtp://hostname[:port][?option=val...]
944 port specifies the RTP port to use.
946 The following URL options are supported:
948 ttl=n
950 Set the TTL (Time-To-Live) value (for multicast only).
951 rtcpport=n
953 Set the remote RTCP port to n.
954 localrtpport=n
956 Set the local RTP port to n.
957 localrtcpport=n'
959 Set the local RTCP port to n.
960 pkt_size=n
962 Set max packet size (in bytes) to n.
963 buffer_size=size
965 Set the maximum UDP socket buffer size in bytes.
966 connect=0|1
968 Do a "connect()" on the UDP socket (if set to 1) or not (if set to
969 0).
970 sources=ip[,ip]
972 List allowed source IP addresses.
973 block=ip[,ip]
975 List disallowed (blocked) source IP addresses.
976 write_to_source=0|1
978 Send packets to the source address of the latest received packet
979 (if set to 1) or to a default remote address (if set to 0).
980 localport=n
982 Set the local RTP port to n.
983 localaddr=addr
985 Local IP address of a network interface used for sending packets or
986 joining multicast groups.
987 timeout=n
989 Set timeout (in microseconds) of socket I/O operations to n.
990 This is a deprecated option. Instead, localrtpport should be used.
992 Important notes:
994 1. If rtcpport is not set the RTCP port will be set to the RTP port
996 value plus 1.
997 2. If localrtpport (the local RTP port) is not set any available port
999 will be used for the local RTP and RTCP ports.
1000 3. If localrtcpport (the local RTCP port) is not set it will be set to
1002 the local RTP port value plus 1.
1003 rtsp
1005 Real-Time Streaming Protocol.
1006 RTSP is not technically a protocol handler in libavformat, it is a
1008 demuxer and muxer. The demuxer supports both normal RTSP (with data
1009 transferred over RTP; this is used by e.g. Apple and Microsoft) and
1010 Real-RTSP (with data transferred over RDT).
1011 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
1013 supporting it (currently Darwin Streaming Server and Mischa
1014 Spiegelmock's <https://github.com/revmischa/rtsp-server>).
1015 The required syntax for a RTSP url is:
1017 rtsp://<hostname>[:<port>]/<path>
1019 Options can be set on the ffmpeg/ffplay command line, or set in code
1021 via "AVOption"s or in "avformat_open_input".
1022 The following options are supported.
1024 initial_pause
1026 Do not start playing the stream immediately if set to 1. Default
1027 value is 0.
1028 rtsp_transport
1030 Set RTSP transport protocols.
1031 It accepts the following values:
1033 udp Use UDP as lower transport protocol.
1035 tcp Use TCP (interleaving within the RTSP control channel) as lower
1037 transport protocol.
1038 udp_multicast
1040 Use UDP multicast as lower transport protocol.
1041 http
1043 Use HTTP tunneling as lower transport protocol, which is useful
1044 for passing proxies.
1045 Multiple lower transport protocols may be specified, in that case
1047 they are tried one at a time (if the setup of one fails, the next
1048 one is tried). For the muxer, only the tcp and udp options are
1049 supported.
1050 rtsp_flags
1052 Set RTSP flags.
1053 The following values are accepted:
1055 filter_src
1057 Accept packets only from negotiated peer address and port.
1058 listen
1060 Act as a server, listening for an incoming connection.
1061 prefer_tcp
1063 Try TCP for RTP transport first, if TCP is available as RTSP
1064 RTP transport.
1065 Default value is none.
1067 allowed_media_types
1069 Set media types to accept from the server.
1070 The following flags are accepted:
1072 video
1074 audio
1075 data
1076 By default it accepts all media types.
1078 min_port
1080 Set minimum local UDP port. Default value is 5000.
1081 max_port
1083 Set maximum local UDP port. Default value is 65000.
1084 listen_timeout
1086 Set maximum timeout (in seconds) to establish an initial
1087 connection. Setting listen_timeout > 0 sets rtsp_flags to listen.
1088 Default is -1 which means an infinite timeout when listen mode is
1089 set.
1090 reorder_queue_size
1092 Set number of packets to buffer for handling of reordered packets.
1093 timeout
1095 Set socket TCP I/O timeout in microseconds.
1096 user_agent
1098 Override User-Agent header. If not specified, it defaults to the
1099 libavformat identifier string.
1100 When receiving data over UDP, the demuxer tries to reorder received
1102 packets (since they may arrive out of order, or packets may get lost
1103 totally). This can be disabled by setting the maximum demuxing delay to
1104 zero (via the "max_delay" field of AVFormatContext).
1105 When watching multi-bitrate Real-RTSP streams with ffplay, the streams
1107 to display can be chosen with "-vst" n and "-ast" n for video and audio
1108 respectively, and can be switched on the fly by pressing "v" and "a".
1109 Examples
1111 The following examples all make use of the ffplay and ffmpeg tools.
1113 • Watch a stream over UDP, with a max reordering delay of 0.5
1115 seconds:
1116 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
1118 • Watch a stream tunneled over HTTP:
1120 ffplay -rtsp_transport http rtsp://server/video.mp4
1122 • Send a stream in realtime to a RTSP server, for others to watch:
1124 ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
1126 • Receive a stream in realtime:
1128 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>
1130 sap
1132 Session Announcement Protocol (RFC 2974). This is not technically a
1133 protocol handler in libavformat, it is a muxer and demuxer. It is used
1134 for signalling of RTP streams, by announcing the SDP for the streams
1135 regularly on a separate port.
1136 Muxer
1138 The syntax for a SAP url given to the muxer is:
1140 sap://<destination>[:<port>][?<options>]
1142 The RTP packets are sent to destination on port port, or to port 5004
1144 if no port is specified. options is a "&"-separated list. The
1145 following options are supported:
1146 announce_addr=address
1148 Specify the destination IP address for sending the announcements
1149 to. If omitted, the announcements are sent to the commonly used
1150 SAP announcement multicast address 224.2.127.254 (sap.mcast.net),
1151 or ff0e::2:7ffe if destination is an IPv6 address.
1152 announce_port=port
1154 Specify the port to send the announcements on, defaults to 9875 if
1155 not specified.
1156 ttl=ttl
1158 Specify the time to live value for the announcements and RTP
1159 packets, defaults to 255.
1160 same_port=0|1
1162 If set to 1, send all RTP streams on the same port pair. If zero
1163 (the default), all streams are sent on unique ports, with each
1164 stream on a port 2 numbers higher than the previous. VLC/Live555
1165 requires this to be set to 1, to be able to receive the stream.
1166 The RTP stack in libavformat for receiving requires all streams to
1167 be sent on unique ports.
1168 Example command lines follow.
1170 To broadcast a stream on the local subnet, for watching in VLC:
1172 ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1
1174 Similarly, for watching in ffplay:
1176 ffmpeg -re -i <input> -f sap sap://224.0.0.255
1178 And for watching in ffplay, over IPv6:
1180 ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]
1182 Demuxer
1184 The syntax for a SAP url given to the demuxer is:
1186 sap://[<address>][:<port>]
1188 address is the multicast address to listen for announcements on, if
1190 omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the
1191 port that is listened on, 9875 if omitted.
1192 The demuxers listens for announcements on the given address and port.
1194 Once an announcement is received, it tries to receive that particular
1195 stream.
1196 Example command lines follow.
1198 To play back the first stream announced on the normal SAP multicast
1200 address:
1201 ffplay sap://
1203 To play back the first stream announced on one the default IPv6 SAP
1205 multicast address:
1206 ffplay sap://[ff0e::2:7ffe]
1208 sctp
1210 Stream Control Transmission Protocol.
1211 The accepted URL syntax is:
1213 sctp://<host>:<port>[?<options>]
1215 The protocol accepts the following options:
1217 listen
1219 If set to any value, listen for an incoming connection. Outgoing
1220 connection is done by default.
1221 max_streams
1223 Set the maximum number of streams. By default no limit is set.
1224 srt
1226 Haivision Secure Reliable Transport Protocol via libsrt.
1227 The supported syntax for a SRT URL is:
1229 srt://<hostname>:<port>[?<options>]
1231 options contains a list of &-separated options of the form key=val.
1233 or
1235 <options> srt://<hostname>:<port>
1237 options contains a list of '-key val' options.
1239 This protocol accepts the following options.
1241 connect_timeout=milliseconds
1243 Connection timeout; SRT cannot connect for RTT > 1500 msec (2
1244 handshake exchanges) with the default connect timeout of 3 seconds.
1245 This option applies to the caller and rendezvous connection modes.
1246 The connect timeout is 10 times the value set for the rendezvous
1247 mode (which can be used as a workaround for this connection problem
1248 with earlier versions).
1249 ffs=bytes
1251 Flight Flag Size (Window Size), in bytes. FFS is actually an
1252 internal parameter and you should set it to not less than
1253 recv_buffer_size and mss. The default value is relatively large,
1254 therefore unless you set a very large receiver buffer, you do not
1255 need to change this option. Default value is 25600.
1256 inputbw=bytes/seconds
1258 Sender nominal input rate, in bytes per seconds. Used along with
1259 oheadbw, when maxbw is set to relative (0), to calculate maximum
1260 sending rate when recovery packets are sent along with the main
1261 media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set
1262 while maxbw is set to relative (0), the actual input rate is
1263 evaluated inside the library. Default value is 0.
1264 iptos=tos
1266 IP Type of Service. Applies to sender only. Default value is 0xB8.
1267 ipttl=ttl
1269 IP Time To Live. Applies to sender only. Default value is 64.
1270 latency=microseconds
1272 Timestamp-based Packet Delivery Delay. Used to absorb bursts of
1273 missed packet retransmissions. This flag sets both rcvlatency and
1274 peerlatency to the same value. Note that prior to version 1.3.0
1275 this is the only flag to set the latency, however this is
1276 effectively equivalent to setting peerlatency, when side is sender
1277 and rcvlatency when side is receiver, and the bidirectional stream
1278 sending is not supported.
1279 listen_timeout=microseconds
1281 Set socket listen timeout.
1282 maxbw=bytes/seconds
1284 Maximum sending bandwidth, in bytes per seconds. -1 infinite
1285 (CSRTCC limit is 30mbps) 0 relative to input rate (see inputbw) >0
1286 absolute limit value Default value is 0 (relative)
1287 mode=caller|listener|rendezvous
1289 Connection mode. caller opens client connection. listener starts
1290 server to listen for incoming connections. rendezvous use Rendez-
1291 Vous connection mode. Default value is caller.
1292 mss=bytes
1294 Maximum Segment Size, in bytes. Used for buffer allocation and rate
1295 calculation using a packet counter assuming fully filled packets.
1296 The smallest MSS between the peers is used. This is 1500 by default
1297 in the overall internet. This is the maximum size of the UDP
1298 packet and can be only decreased, unless you have some unusual
1299 dedicated network settings. Default value is 1500.
1300 nakreport=1|0
1302 If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
1303 periodically until a lost packet is retransmitted or intentionally
1304 dropped. Default value is 1.
1305 oheadbw=percents
1307 Recovery bandwidth overhead above input rate, in percents. See
1308 inputbw. Default value is 25%.
1309 passphrase=string
1311 HaiCrypt Encryption/Decryption Passphrase string, length from 10 to
1312 79 characters. The passphrase is the shared secret between the
1313 sender and the receiver. It is used to generate the Key Encrypting
1314 Key using PBKDF2 (Password-Based Key Derivation Function). It is
1315 used only if pbkeylen is non-zero. It is used on the receiver only
1316 if the received data is encrypted. The configured passphrase
1317 cannot be recovered (write-only).
1318 enforced_encryption=1|0
1320 If true, both connection parties must have the same password set
1321 (including empty, that is, with no encryption). If the password
1322 doesn't match or only one side is unencrypted, the connection is
1323 rejected. Default is true.
1324 kmrefreshrate=packets
1326 The number of packets to be transmitted after which the encryption
1327 key is switched to a new key. Default is -1. -1 means auto
1328 (0x1000000 in srt library). The range for this option is integers
1329 in the 0 - "INT_MAX".
1330 kmpreannounce=packets
1332 The interval between when a new encryption key is sent and when
1333 switchover occurs. This value also applies to the subsequent
1334 interval between when switchover occurs and when the old encryption
1335 key is decommissioned. Default is -1. -1 means auto (0x1000 in srt
1336 library). The range for this option is integers in the 0 -
1337 "INT_MAX".
1338 snddropdelay=microseconds
1340 The sender's extra delay before dropping packets. This delay is
1341 added to the default drop delay time interval value.
1342 Special value -1: Do not drop packets on the sender at all.
1344 payload_size=bytes
1346 Sets the maximum declared size of a packet transferred during the
1347 single call to the sending function in Live mode. Use 0 if this
1348 value isn't used (which is default in file mode). Default is -1
1349 (automatic), which typically means MPEG-TS; if you are going to use
1350 SRT to send any different kind of payload, such as, for example,
1351 wrapping a live stream in very small frames, then you can use a
1352 bigger maximum frame size, though not greater than 1456 bytes.
1353 pkt_size=bytes
1355 Alias for payload_size.
1356 peerlatency=microseconds
1358 The latency value (as described in rcvlatency) that is set by the
1359 sender side as a minimum value for the receiver.
1360 pbkeylen=bytes
1362 Sender encryption key length, in bytes. Only can be set to 0, 16,
1363 24 and 32. Enable sender encryption if not 0. Not required on
1364 receiver (set to 0), key size obtained from sender in HaiCrypt
1365 handshake. Default value is 0.
1366 rcvlatency=microseconds
1368 The time that should elapse since the moment when the packet was
1369 sent and the moment when it's delivered to the receiver application
1370 in the receiving function. This time should be a buffer time large
1371 enough to cover the time spent for sending, unexpectedly extended
1372 RTT time, and the time needed to retransmit the lost UDP packet.
1373 The effective latency value will be the maximum of this options'
1374 value and the value of peerlatency set by the peer side. Before
1375 version 1.3.0 this option is only available as latency.
1376 recv_buffer_size=bytes
1378 Set UDP receive buffer size, expressed in bytes.
1379 send_buffer_size=bytes
1381 Set UDP send buffer size, expressed in bytes.
1382 timeout=microseconds
1384 Set raise error timeouts for read, write and connect operations.
1385 Note that the SRT library has internal timeouts which can be
1386 controlled separately, the value set here is only a cap on those.
1387 tlpktdrop=1|0
1389 Too-late Packet Drop. When enabled on receiver, it skips missing
1390 packets that have not been delivered in time and delivers the
1391 following packets to the application when their time-to-play has
1392 come. It also sends a fake ACK to the sender. When enabled on
1393 sender and enabled on the receiving peer, the sender drops the
1394 older packets that have no chance of being delivered in time. It
1395 was automatically enabled in the sender if the receiver supports
1396 it.
1397 sndbuf=bytes
1399 Set send buffer size, expressed in bytes.
1400 rcvbuf=bytes
1402 Set receive buffer size, expressed in bytes.
1403 Receive buffer must not be greater than ffs.
1405 lossmaxttl=packets
1407 The value up to which the Reorder Tolerance may grow. When Reorder
1408 Tolerance is > 0, then packet loss report is delayed until that
1409 number of packets come in. Reorder Tolerance increases every time a
1410 "belated" packet has come, but it wasn't due to retransmission
1411 (that is, when UDP packets tend to come out of order), with the
1412 difference between the latest sequence and this packet's sequence,
1413 and not more than the value of this option. By default it's 0,
1414 which means that this mechanism is turned off, and the loss report
1415 is always sent immediately upon experiencing a "gap" in sequences.
1416 minversion
1418 The minimum SRT version that is required from the peer. A
1419 connection to a peer that does not satisfy the minimum version
1420 requirement will be rejected.
1421 The version format in hex is 0xXXYYZZ for x.y.z in human readable
1423 form.
1424 streamid=string
1426 A string limited to 512 characters that can be set on the socket
1427 prior to connecting. This stream ID will be able to be retrieved by
1428 the listener side from the socket that is returned from srt_accept
1429 and was connected by a socket with that set stream ID. SRT does not
1430 enforce any special interpretation of the contents of this string.
1431 This option doesnXt make sense in Rendezvous connection; the result
1432 might be that simply one side will override the value from the
1433 other side and itXs the matter of luck which one would win
1434 srt_streamid=string
1436 Alias for streamid to avoid conflict with ffmpeg command line
1437 option.
1438 smoother=live|file
1440 The type of Smoother used for the transmission for that socket,
1441 which is responsible for the transmission and congestion control.
1442 The Smoother type must be exactly the same on both connecting
1443 parties, otherwise the connection is rejected.
1444 messageapi=1|0
1446 When set, this socket uses the Message API, otherwise it uses
1447 Buffer API. Note that in live mode (see transtype) thereXs only
1448 message API available. In File mode you can chose to use one of two
1449 modes:
1450 Stream API (default, when this option is false). In this mode you
1452 may send as many data as you wish with one sending instruction, or
1453 even use dedicated functions that read directly from a file. The
1454 internal facility will take care of any speed and congestion
1455 control. When receiving, you can also receive as many data as
1456 desired, the data not extracted will be waiting for the next call.
1457 There is no boundary between data portions in the Stream mode.
1458 Message API. In this mode your single sending instruction passes
1460 exactly one piece of data that has boundaries (a message). Contrary
1461 to Live mode, this message may span across multiple UDP packets and
1462 the only size limitation is that it shall fit as a whole in the
1463 sending buffer. The receiver shall use as large buffer as necessary
1464 to receive the message, otherwise the message will not be given up.
1465 When the message is not complete (not all packets received or there
1466 was a packet loss) it will not be given up.
1467 transtype=live|file
1469 Sets the transmission type for the socket, in particular, setting
1470 this option sets multiple other parameters to their default values
1471 as required for a particular transmission type.
1472 live: Set options as for live transmission. In this mode, you
1474 should send by one sending instruction only so many data that fit
1475 in one UDP packet, and limited to the value defined first in
1476 payload_size (1316 is default in this mode). There is no speed
1477 control in this mode, only the bandwidth control, if configured, in
1478 order to not exceed the bandwidth with the overhead transmission
1479 (retransmitted and control packets).
1480 file: Set options as for non-live transmission. See messageapi for
1482 further explanations
1483 linger=seconds
1485 The number of seconds that the socket waits for unsent data when
1486 closing. Default is -1. -1 means auto (off with 0 seconds in live
1487 mode, on with 180 seconds in file mode). The range for this option
1488 is integers in the 0 - "INT_MAX".
1489 tsbpd=1|0
1491 When true, use Timestamp-based Packet Delivery mode. The default
1492 behavior depends on the transmission type: enabled in live mode,
1493 disabled in file mode.
1494 For more information see: <https://github.com/Haivision/srt>.
1496 srtp
1498 Secure Real-time Transport Protocol.
1499 The accepted options are:
1501 srtp_in_suite
1503 srtp_out_suite
1504 Select input and output encoding suites.
1505 Supported values:
1507 AES_CM_128_HMAC_SHA1_80
1509 SRTP_AES128_CM_HMAC_SHA1_80
1510 AES_CM_128_HMAC_SHA1_32
1511 SRTP_AES128_CM_HMAC_SHA1_32
1512 srtp_in_params
1513 srtp_out_params
1514 Set input and output encoding parameters, which are expressed by a
1515 base64-encoded representation of a binary block. The first 16 bytes
1516 of this binary block are used as master key, the following 14 bytes
1517 are used as master salt.
1518 subfile
1520 Virtually extract a segment of a file or another stream. The
1521 underlying stream must be seekable.
1522 Accepted options:
1524 start
1526 Start offset of the extracted segment, in bytes.
1527 end End offset of the extracted segment, in bytes. If set to 0,
1529 extract till end of file.
1530 Examples:
1532 Extract a chapter from a DVD VOB file (start and end sectors obtained
1534 externally and multiplied by 2048):
1535 subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
1537 Play an AVI file directly from a TAR archive:
1539 subfile,,start,183241728,end,366490624,,:archive.tar
1541 Play a MPEG-TS file from start offset till end:
1543 subfile,,start,32815239,end,0,,:video.ts
1545 tee
1547 Writes the output to multiple protocols. The individual outputs are
1548 separated by |
1549 tee:file://path/to/local/this.avi|file://path/to/local/that.avi
1551 tcp
1553 Transmission Control Protocol.
1554 The required syntax for a TCP url is:
1556 tcp://<hostname>:<port>[?<options>]
1558 options contains a list of &-separated options of the form key=val.
1560 The list of supported options follows.
1562 listen=2|1|0
1564 Listen for an incoming connection. 0 disables listen, 1 enables
1565 listen in single client mode, 2 enables listen in multi-client
1566 mode. Default value is 0.
1567 timeout=microseconds
1569 Set raise error timeout, expressed in microseconds.
1570 This option is only relevant in read mode: if no data arrived in
1572 more than this time interval, raise error.
1573 listen_timeout=milliseconds
1575 Set listen timeout, expressed in milliseconds.
1576 recv_buffer_size=bytes
1578 Set receive buffer size, expressed bytes.
1579 send_buffer_size=bytes
1581 Set send buffer size, expressed bytes.
1582 tcp_nodelay=1|0
1584 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
1585 Remark: Writing to the socket is currently not optimized to
1587 minimize system calls and reduces the efficiency / effect of
1588 TCP_NODELAY.
1589 tcp_mss=bytes
1591 Set maximum segment size for outgoing TCP packets, expressed in
1592 bytes.
1593 The following example shows how to setup a listening TCP connection
1595 with ffmpeg, which is then accessed with ffplay:
1596 ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
1598 ffplay tcp://<hostname>:<port>
1599 tls
1601 Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
1602 The required syntax for a TLS/SSL url is:
1604 tls://<hostname>:<port>[?<options>]
1606 The following parameters can be set via command line options (or in
1608 code via "AVOption"s):
1609 ca_file, cafile=filename
1611 A file containing certificate authority (CA) root certificates to
1612 treat as trusted. If the linked TLS library contains a default this
1613 might not need to be specified for verification to work, but not
1614 all libraries and setups have defaults built in. The file must be
1615 in OpenSSL PEM format.
1616 tls_verify=1|0
1618 If enabled, try to verify the peer that we are communicating with.
1619 Note, if using OpenSSL, this currently only makes sure that the
1620 peer certificate is signed by one of the root certificates in the
1621 CA database, but it does not validate that the certificate actually
1622 matches the host name we are trying to connect to. (With other
1623 backends, the host name is validated as well.)
1624 This is disabled by default since it requires a CA database to be
1626 provided by the caller in many cases.
1627 cert_file, cert=filename
1629 A file containing a certificate to use in the handshake with the
1630 peer. (When operating as server, in listen mode, this is more
1631 often required by the peer, while client certificates only are
1632 mandated in certain setups.)
1633 key_file, key=filename
1635 A file containing the private key for the certificate.
1636 listen=1|0
1638 If enabled, listen for connections on the provided port, and assume
1639 the server role in the handshake instead of the client role.
1640 http_proxy
1642 The HTTP proxy to tunnel through, e.g. "http://example.com:1234".
1643 The proxy must support the CONNECT method.
1644 Example command lines:
1646 To create a TLS/SSL server that serves an input stream.
1648 ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>
1650 To play back a stream from the TLS/SSL server using ffplay:
1652 ffplay tls://<hostname>:<port>
1654 udp
1656 User Datagram Protocol.
1657 The required syntax for an UDP URL is:
1659 udp://<hostname>:<port>[?<options>]
1661 options contains a list of &-separated options of the form key=val.
1663 In case threading is enabled on the system, a circular buffer is used
1665 to store the incoming data, which allows one to reduce loss of data due
1666 to UDP socket buffer overruns. The fifo_size and overrun_nonfatal
1667 options are related to this buffer.
1668 The list of supported options follows.
1670 buffer_size=size
1672 Set the UDP maximum socket buffer size in bytes. This is used to
1673 set either the receive or send buffer size, depending on what the
1674 socket is used for. Default is 32 KB for output, 384 KB for input.
1675 See also fifo_size.
1676 bitrate=bitrate
1678 If set to nonzero, the output will have the specified constant
1679 bitrate if the input has enough packets to sustain it.
1680 burst_bits=bits
1682 When using bitrate this specifies the maximum number of bits in
1683 packet bursts.
1684 localport=port
1686 Override the local UDP port to bind with.
1687 localaddr=addr
1689 Local IP address of a network interface used for sending packets or
1690 joining multicast groups.
1691 pkt_size=size
1693 Set the size in bytes of UDP packets.
1694 reuse=1|0
1696 Explicitly allow or disallow reusing UDP sockets.
1697 ttl=ttl
1699 Set the time to live value (for multicast only).
1700 connect=1|0
1702 Initialize the UDP socket with "connect()". In this case, the
1703 destination address can't be changed with ff_udp_set_remote_url
1704 later. If the destination address isn't known at the start, this
1705 option can be specified in ff_udp_set_remote_url, too. This allows
1706 finding out the source address for the packets with getsockname,
1707 and makes writes return with AVERROR(ECONNREFUSED) if "destination
1708 unreachable" is received. For receiving, this gives the benefit of
1709 only receiving packets from the specified peer address/port.
1710 sources=address[,address]
1712 Only receive packets sent from the specified addresses. In case of
1713 multicast, also subscribe to multicast traffic coming from these
1714 addresses only.
1715 block=address[,address]
1717 Ignore packets sent from the specified addresses. In case of
1718 multicast, also exclude the source addresses in the multicast
1719 subscription.
1720 fifo_size=units
1722 Set the UDP receiving circular buffer size, expressed as a number
1723 of packets with size of 188 bytes. If not specified defaults to
1724 7*4096.
1725 overrun_nonfatal=1|0
1727 Survive in case of UDP receiving circular buffer overrun. Default
1728 value is 0.
1729 timeout=microseconds
1731 Set raise error timeout, expressed in microseconds.
1732 This option is only relevant in read mode: if no data arrived in
1734 more than this time interval, raise error.
1735 broadcast=1|0
1737 Explicitly allow or disallow UDP broadcasting.
1738 Note that broadcasting may not work properly on networks having a
1740 broadcast storm protection.
1741 Examples
1743 • Use ffmpeg to stream over UDP to a remote endpoint:
1745 ffmpeg -i <input> -f <format> udp://<hostname>:<port>
1747 • Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP
1749 packets, using a large input buffer:
1750 ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535
1752 • Use ffmpeg to receive over UDP from a remote endpoint:
1754 ffmpeg -i udp://[<multicast-address>]:<port> ...
1756 unix
1758 Unix local socket
1759 The required syntax for a Unix socket URL is:
1761 unix://<filepath>
1763 The following parameters can be set via command line options (or in
1765 code via "AVOption"s):
1766 timeout
1768 Timeout in ms.
1769 listen
1771 Create the Unix socket in listening mode.
1772 zmq
1774 ZeroMQ asynchronous messaging using the libzmq library.
1775 This library supports unicast streaming to multiple clients without
1777 relying on an external server.
1778 The required syntax for streaming or connecting to a stream is:
1780 zmq:tcp://ip-address:port
1782 Example: Create a localhost stream on port 5555:
1784 ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
1786 Multiple clients may connect to the stream using:
1788 ffplay zmq:tcp://127.0.0.1:5555
1790 Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub
1792 pattern. The server side binds to a port and publishes data. Clients
1793 connect to the server (via IP address/port) and subscribe to the
1794 stream. The order in which the server and client start generally does
1795 not matter.
1796 ffmpeg must be compiled with the --enable-libzmq option to support this
1798 protocol.
1799 Options can be set on the ffmpeg/ffplay command line. The following
1801 options are supported:
1802 pkt_size
1804 Forces the maximum packet size for sending/receiving data. The
1805 default value is 131,072 bytes. On the server side, this sets the
1806 maximum size of sent packets via ZeroMQ. On the clients, it sets an
1807 internal buffer size for receiving packets. Note that pkt_size on
1808 the clients should be equal to or greater than pkt_size on the
1809 server. Otherwise the received message may be truncated causing
1810 decoding errors.
1811
1813 ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
1814
1816 The FFmpeg developers.
1817 For details about the authorship, see the Git history of the project
1819 (git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
1820 the FFmpeg source directory, or browsing the online repository at
1821 <http://source.ffmpeg.org>.
1822 Maintainers for the specific components are listed in the file
1824 MAINTAINERS in the source code tree.
1825 FFMPEG-PROTOCOLS(1)