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 @file{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{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.).
185 It is the path or name of the resource to play with reference to the
186 application specified in @var{app}, may be prefixed by "mp4:".
190 For example to read with @file{avplay} a multimedia resource named
191 "sample" from the application "vod" from an RTMP server "myserver":
193 avplay rtmp://myserver/vod/sample
196 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
198 Real-Time Messaging Protocol and its variants supported through
201 Requires the presence of the librtmp headers and library during
202 configuration. You need to explicitly configure the build with
203 "--enable-librtmp". If enabled this will replace the native RTMP
206 This protocol provides most client functions and a few server
207 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
208 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
209 variants of these encrypted types (RTMPTE, RTMPTS).
211 The required syntax is:
213 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
216 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
217 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
218 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
219 meaning as specified for the RTMP native protocol.
220 @var{options} contains a list of space-separated options of the form
223 See the librtmp manual page (man 3 librtmp) for more information.
225 For example, to stream a file in real-time to an RTMP server using
228 avconv -re -i myfile -f flv rtmp://myserver/live/mystream
231 To play the same stream using @file{avplay}:
233 avplay "rtmp://myserver/live/mystream live=1"
242 RTSP is not technically a protocol handler in libavformat, it is a demuxer
243 and muxer. The demuxer supports both normal RTSP (with data transferred
244 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
245 data transferred over RDT).
247 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
248 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
249 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
251 The required syntax for a RTSP url is:
253 rtsp://@var{hostname}[:@var{port}]/@var{path}
256 The following options (set on the @command{avconv}/@file{avplay} command
257 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
260 Flags for @code{rtsp_transport}:
265 Use UDP as lower transport protocol.
268 Use TCP (interleaving within the RTSP control channel) as lower
272 Use UDP multicast as lower transport protocol.
275 Use HTTP tunneling as lower transport protocol, which is useful for
279 Multiple lower transport protocols may be specified, in that case they are
280 tried one at a time (if the setup of one fails, the next one is tried).
281 For the muxer, only the @code{tcp} and @code{udp} options are supported.
283 Flags for @code{rtsp_flags}:
287 Accept packets only from negotiated peer address and port.
290 When receiving data over UDP, the demuxer tries to reorder received packets
291 (since they may arrive out of order, or packets may get lost totally). In
292 order for this to be enabled, a maximum delay must be specified in the
293 @code{max_delay} field of AVFormatContext.
295 When watching multi-bitrate Real-RTSP streams with @file{avplay}, the
296 streams to display can be chosen with @code{-vst} @var{n} and
297 @code{-ast} @var{n} for video and audio respectively, and can be switched
298 on the fly by pressing @code{v} and @code{a}.
300 Example command lines:
302 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
305 avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
308 To watch a stream tunneled over HTTP:
311 avplay -rtsp_transport http rtsp://server/video.mp4
314 To send a stream in realtime to a RTSP server, for others to watch:
317 avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
322 Session Announcement Protocol (RFC 2974). This is not technically a
323 protocol handler in libavformat, it is a muxer and demuxer.
324 It is used for signalling of RTP streams, by announcing the SDP for the
325 streams regularly on a separate port.
329 The syntax for a SAP url given to the muxer is:
331 sap://@var{destination}[:@var{port}][?@var{options}]
334 The RTP packets are sent to @var{destination} on port @var{port},
335 or to port 5004 if no port is specified.
336 @var{options} is a @code{&}-separated list. The following options
341 @item announce_addr=@var{address}
342 Specify the destination IP address for sending the announcements to.
343 If omitted, the announcements are sent to the commonly used SAP
344 announcement multicast address 224.2.127.254 (sap.mcast.net), or
345 ff0e::2:7ffe if @var{destination} is an IPv6 address.
347 @item announce_port=@var{port}
348 Specify the port to send the announcements on, defaults to
349 9875 if not specified.
352 Specify the time to live value for the announcements and RTP packets,
355 @item same_port=@var{0|1}
356 If set to 1, send all RTP streams on the same port pair. If zero (the
357 default), all streams are sent on unique ports, with each stream on a
358 port 2 numbers higher than the previous.
359 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
360 The RTP stack in libavformat for receiving requires all streams to be sent
364 Example command lines follow.
366 To broadcast a stream on the local subnet, for watching in VLC:
369 avconv -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
372 Similarly, for watching in avplay:
375 avconv -re -i @var{input} -f sap sap://224.0.0.255
378 And for watching in avplay, over IPv6:
381 avconv -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
386 The syntax for a SAP url given to the demuxer is:
388 sap://[@var{address}][:@var{port}]
391 @var{address} is the multicast address to listen for announcements on,
392 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
393 is the port that is listened on, 9875 if omitted.
395 The demuxers listens for announcements on the given address and port.
396 Once an announcement is received, it tries to receive that particular stream.
398 Example command lines follow.
400 To play back the first stream announced on the normal SAP multicast address:
406 To play back the first stream announced on one the default IPv6 SAP multicast address:
409 avplay sap://[ff0e::2:7ffe]
414 Trasmission Control Protocol.
416 The required syntax for a TCP url is:
418 tcp://@var{hostname}:@var{port}[?@var{options}]
424 Listen for an incoming connection
427 avconv -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
428 avplay tcp://@var{hostname}:@var{port}
435 User Datagram Protocol.
437 The required syntax for a UDP url is:
439 udp://@var{hostname}:@var{port}[?@var{options}]
442 @var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
443 Follow the list of supported options.
447 @item buffer_size=@var{size}
448 set the UDP buffer size in bytes
450 @item localport=@var{port}
451 override the local UDP port to bind with
453 @item localaddr=@var{addr}
454 Choose the local IP address. This is useful e.g. if sending multicast
455 and the host has multiple interfaces, where the user can choose
456 which interface to send on by specifying the IP address of that interface.
458 @item pkt_size=@var{size}
459 set the size in bytes of UDP packets
461 @item reuse=@var{1|0}
462 explicitly allow or disallow reusing UDP sockets
465 set the time to live value (for multicast only)
467 @item connect=@var{1|0}
468 Initialize the UDP socket with @code{connect()}. In this case, the
469 destination address can't be changed with ff_udp_set_remote_url later.
470 If the destination address isn't known at the start, this option can
471 be specified in ff_udp_set_remote_url, too.
472 This allows finding out the source address for the packets with getsockname,
473 and makes writes return with AVERROR(ECONNREFUSED) if "destination
474 unreachable" is received.
475 For receiving, this gives the benefit of only receiving packets from
476 the specified peer address/port.
479 Some usage examples of the udp protocol with @command{avconv} follow.
481 To stream over UDP to a remote endpoint:
483 avconv -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
486 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
488 avconv -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
491 To receive over UDP from a remote endpoint:
493 avconv -i udp://[@var{multicast-address}]:@var{port}