4 Protocols are configured elements in Libav which allow to access
5 resources which require the use of a particular protocol.
7 When you configure your Libav build, all the supported protocols are
8 enabled by default. You can list all available ones using the
9 configure option "--list-protocols".
11 You can disable all the protocols using the configure option
12 "--disable-protocols", and selectively enable a protocol using the
13 option "--enable-protocol=@var{PROTOCOL}", or you can disable a
14 particular protocol using the option
15 "--disable-protocol=@var{PROTOCOL}".
17 The option "-protocols" of the ff* tools will display the list of
20 A description of the currently available protocols follows.
24 Physical concatenation protocol.
26 Allow to read and seek from many resource in sequence as if they were
29 A URL accepted by this protocol has the syntax:
31 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
34 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
35 resource to be concatenated, each one possibly specifying a distinct
38 For example to read a sequence of files @file{split1.mpeg},
39 @file{split2.mpeg}, @file{split3.mpeg} with @command{avplay} use the
42 avplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
45 Note that you may need to escape the character "|" which is special for
52 Allow to read from or read to a file.
54 For example to read from a file @file{input.mpeg} with @command{avconv}
57 avconv -i file:input.mpeg output.mpeg
60 The ff* tools default to the file protocol, that is a resource
61 specified with the name "FILE.mpeg" is interpreted as the URL
70 Read Apple HTTP Live Streaming compliant segmented stream as
71 a uniform one. The M3U8 playlists describing the segments can be
72 remote HTTP resources or local files, accessed using the standard
74 The nested protocol is declared by specifying
75 "+@var{proto}" after the hls URI scheme name, where @var{proto}
76 is either "file" or "http".
79 hls+http://host/path/to/remote/resource.m3u8
80 hls+file://path/to/local/resource.m3u8
83 Using this protocol is discouraged - the hls demuxer should work
84 just as well (if not, please report the issues) and is more complete.
85 To use the hls demuxer instead, simply use the direct URLs to the
90 HTTP (Hyper Text Transfer Protocol).
94 MMS (Microsoft Media Server) protocol over TCP.
98 MMS (Microsoft Media Server) protocol over HTTP.
100 The required syntax is:
102 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
109 Computes the MD5 hash of the data to be written, and on close writes
110 this to the designated output or stdout if none is specified. It can
111 be used to test muxers without writing an actual file.
113 Some examples follow.
115 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
116 avconv -i input.flv -f avi -y md5:output.avi.md5
118 # Write the MD5 hash of the encoded AVI file to stdout.
119 avconv -i input.flv -f avi -y md5:
122 Note that some formats (typically MOV) require the output protocol to
123 be seekable, so they will fail with the MD5 output protocol.
127 UNIX pipe access protocol.
129 Allow to read and write from UNIX pipes.
131 The accepted syntax is:
136 @var{number} is the number corresponding to the file descriptor of the
137 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
138 is not specified, by default the stdout file descriptor will be used
139 for writing, stdin for reading.
141 For example to read from stdin with @command{avconv}:
143 cat test.wav | avconv -i pipe:0
144 # ...this is the same as...
145 cat test.wav | avconv -i pipe:
148 For writing to stdout with @command{avconv}:
150 avconv -i test.wav -f avi pipe:1 | cat > test.avi
151 # ...this is the same as...
152 avconv -i test.wav -f avi pipe: | cat > test.avi
155 Note that some formats (typically MOV), require the output protocol to
156 be seekable, so they will fail with the pipe output protocol.
160 Real-Time Messaging Protocol.
162 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
163 content across a TCP/IP network.
165 The required syntax is:
167 rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
170 The accepted parameters are:
174 The address of the RTMP server.
177 The number of the TCP port to use (by default is 1935).
180 It is the name of the application to access. It usually corresponds to
181 the path where the application is installed on the RTMP server
182 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
183 the value parsed from the URI through the @code{rtmp_app} option, too.
186 It is the path or name of the resource to play with reference to the
187 application specified in @var{app}, may be prefixed by "mp4:". You
188 can override the value parsed from the URI through the @code{rtmp_playpath}
193 Additionally, the following parameters can be set via command line options
194 (or in code via @code{AVOption}s):
198 Name of application to connect on the RTMP server. This option
199 overrides the parameter specified in the URI.
202 Set the client buffer time in milliseconds. The default is 3000.
205 Extra arbitrary AMF connection parameters, parsed from a string,
206 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
207 Each value is prefixed by a single character denoting the type,
208 B for Boolean, N for number, S for string, O for object, or Z for null,
209 followed by a colon. For Booleans the data must be either 0 or 1 for
210 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
211 1 to end or begin an object, respectively. Data items in subobjects may
212 be named, by prefixing the type with 'N' and specifying the name before
213 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
214 times to construct arbitrary AMF sequences.
217 Version of the Flash plugin used to run the SWF player. The default
220 @item rtmp_flush_interval
221 Number of packets flushed in the same request (RTMPT only). The default
225 Specify that the media is a live stream. No resuming or seeking in
226 live streams is possible. The default value is @code{any}, which means the
227 subscriber first tries to play the live stream specified in the
228 playpath. If a live stream of that name is not found, it plays the
229 recorded stream. The other possible values are @code{live} and
233 Stream identifier to play or to publish. This option overrides the
234 parameter specified in the URI.
237 URL of the SWF player for the media. By default no value will be sent.
240 URL of the target stream. Defaults to proto://host[:port]/app.
244 For example to read with @command{avplay} a multimedia resource named
245 "sample" from the application "vod" from an RTMP server "myserver":
247 avplay rtmp://myserver/vod/sample
252 Encrypted Real-Time Messaging Protocol.
254 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
255 streaming multimedia content within standard cryptographic primitives,
256 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
261 Real-Time Messaging Protocol over a secure SSL connection.
263 The Real-Time Messaging Protocol (RTMPS) is used for streaming
264 multimedia content across an encrypted connection.
268 Real-Time Messaging Protocol tunneled through HTTP.
270 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
271 for streaming multimedia content within HTTP requests to traverse
276 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
278 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
279 is used for streaming multimedia content within HTTP requests to traverse
284 Real-Time Messaging Protocol tunneled through HTTPS.
286 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
287 for streaming multimedia content within HTTPS requests to traverse
290 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
292 Real-Time Messaging Protocol and its variants supported through
295 Requires the presence of the librtmp headers and library during
296 configuration. You need to explicitly configure the build with
297 "--enable-librtmp". If enabled this will replace the native RTMP
300 This protocol provides most client functions and a few server
301 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
302 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
303 variants of these encrypted types (RTMPTE, RTMPTS).
305 The required syntax is:
307 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
310 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
311 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
312 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
313 meaning as specified for the RTMP native protocol.
314 @var{options} contains a list of space-separated options of the form
317 See the librtmp manual page (man 3 librtmp) for more information.
319 For example, to stream a file in real-time to an RTMP server using
322 avconv -re -i myfile -f flv rtmp://myserver/live/mystream
325 To play the same stream using @command{avplay}:
327 avplay "rtmp://myserver/live/mystream live=1"
336 RTSP is not technically a protocol handler in libavformat, it is a demuxer
337 and muxer. The demuxer supports both normal RTSP (with data transferred
338 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
339 data transferred over RDT).
341 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
342 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
343 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
345 The required syntax for a RTSP url is:
347 rtsp://@var{hostname}[:@var{port}]/@var{path}
350 The following options (set on the @command{avconv}/@command{avplay} command
351 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
354 Flags for @code{rtsp_transport}:
359 Use UDP as lower transport protocol.
362 Use TCP (interleaving within the RTSP control channel) as lower
366 Use UDP multicast as lower transport protocol.
369 Use HTTP tunneling as lower transport protocol, which is useful for
373 Multiple lower transport protocols may be specified, in that case they are
374 tried one at a time (if the setup of one fails, the next one is tried).
375 For the muxer, only the @code{tcp} and @code{udp} options are supported.
377 Flags for @code{rtsp_flags}:
381 Accept packets only from negotiated peer address and port.
383 Act as a server, listening for an incoming connection.
386 When receiving data over UDP, the demuxer tries to reorder received packets
387 (since they may arrive out of order, or packets may get lost totally). This
388 can be disabled by setting the maximum demuxing delay to zero (via
389 the @code{max_delay} field of AVFormatContext).
391 When watching multi-bitrate Real-RTSP streams with @command{avplay}, the
392 streams to display can be chosen with @code{-vst} @var{n} and
393 @code{-ast} @var{n} for video and audio respectively, and can be switched
394 on the fly by pressing @code{v} and @code{a}.
396 Example command lines:
398 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
401 avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
404 To watch a stream tunneled over HTTP:
407 avplay -rtsp_transport http rtsp://server/video.mp4
410 To send a stream in realtime to a RTSP server, for others to watch:
413 avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
416 To receive a stream in realtime:
419 avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
424 Session Announcement Protocol (RFC 2974). This is not technically a
425 protocol handler in libavformat, it is a muxer and demuxer.
426 It is used for signalling of RTP streams, by announcing the SDP for the
427 streams regularly on a separate port.
431 The syntax for a SAP url given to the muxer is:
433 sap://@var{destination}[:@var{port}][?@var{options}]
436 The RTP packets are sent to @var{destination} on port @var{port},
437 or to port 5004 if no port is specified.
438 @var{options} is a @code{&}-separated list. The following options
443 @item announce_addr=@var{address}
444 Specify the destination IP address for sending the announcements to.
445 If omitted, the announcements are sent to the commonly used SAP
446 announcement multicast address 224.2.127.254 (sap.mcast.net), or
447 ff0e::2:7ffe if @var{destination} is an IPv6 address.
449 @item announce_port=@var{port}
450 Specify the port to send the announcements on, defaults to
451 9875 if not specified.
454 Specify the time to live value for the announcements and RTP packets,
457 @item same_port=@var{0|1}
458 If set to 1, send all RTP streams on the same port pair. If zero (the
459 default), all streams are sent on unique ports, with each stream on a
460 port 2 numbers higher than the previous.
461 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
462 The RTP stack in libavformat for receiving requires all streams to be sent
466 Example command lines follow.
468 To broadcast a stream on the local subnet, for watching in VLC:
471 avconv -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
474 Similarly, for watching in avplay:
477 avconv -re -i @var{input} -f sap sap://224.0.0.255
480 And for watching in avplay, over IPv6:
483 avconv -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
488 The syntax for a SAP url given to the demuxer is:
490 sap://[@var{address}][:@var{port}]
493 @var{address} is the multicast address to listen for announcements on,
494 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
495 is the port that is listened on, 9875 if omitted.
497 The demuxers listens for announcements on the given address and port.
498 Once an announcement is received, it tries to receive that particular stream.
500 Example command lines follow.
502 To play back the first stream announced on the normal SAP multicast address:
508 To play back the first stream announced on one the default IPv6 SAP multicast address:
511 avplay sap://[ff0e::2:7ffe]
516 Trasmission Control Protocol.
518 The required syntax for a TCP url is:
520 tcp://@var{hostname}:@var{port}[?@var{options}]
526 Listen for an incoming connection
529 avconv -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
530 avplay tcp://@var{hostname}:@var{port}
537 User Datagram Protocol.
539 The required syntax for a UDP url is:
541 udp://@var{hostname}:@var{port}[?@var{options}]
544 @var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
545 Follow the list of supported options.
549 @item buffer_size=@var{size}
550 set the UDP buffer size in bytes
552 @item localport=@var{port}
553 override the local UDP port to bind with
555 @item localaddr=@var{addr}
556 Choose the local IP address. This is useful e.g. if sending multicast
557 and the host has multiple interfaces, where the user can choose
558 which interface to send on by specifying the IP address of that interface.
560 @item pkt_size=@var{size}
561 set the size in bytes of UDP packets
563 @item reuse=@var{1|0}
564 explicitly allow or disallow reusing UDP sockets
567 set the time to live value (for multicast only)
569 @item connect=@var{1|0}
570 Initialize the UDP socket with @code{connect()}. In this case, the
571 destination address can't be changed with ff_udp_set_remote_url later.
572 If the destination address isn't known at the start, this option can
573 be specified in ff_udp_set_remote_url, too.
574 This allows finding out the source address for the packets with getsockname,
575 and makes writes return with AVERROR(ECONNREFUSED) if "destination
576 unreachable" is received.
577 For receiving, this gives the benefit of only receiving packets from
578 the specified peer address/port.
580 @item sources=@var{address}[,@var{address}]
581 Only receive packets sent to the multicast group from one of the
582 specified sender IP addresses.
584 @item block=@var{address}[,@var{address}]
585 Ignore packets sent to the multicast group from the specified
589 Some usage examples of the udp protocol with @command{avconv} follow.
591 To stream over UDP to a remote endpoint:
593 avconv -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
596 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
598 avconv -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
601 To receive over UDP from a remote endpoint:
603 avconv -i udp://[@var{multicast-address}]:@var{port}