4 Protocols are configured elements in FFmpeg which allow to access
5 resources which require the use of a particular protocol.
7 When you configure your FFmpeg 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.
26 The accepted options are:
36 Playlist to read (BDMV/PLAYLIST/?????.mpls)
42 Read longest playlist from BluRay mounted to /mnt/bluray:
47 Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
49 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
54 Caching wrapper for input stream.
56 Cache the input stream to temporary file. It brings seeking capability to live streams.
64 Physical concatenation protocol.
66 Allow to read and seek from many resource in sequence as if they were
69 A URL accepted by this protocol has the syntax:
71 concat:@var{URL1}|@var{URL2}|...|@var{URLN}
74 where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
75 resource to be concatenated, each one possibly specifying a distinct
78 For example to read a sequence of files @file{split1.mpeg},
79 @file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
82 ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
85 Note that you may need to escape the character "|" which is special for
90 AES-encrypted stream reading protocol.
92 The accepted options are:
95 Set the AES decryption key binary block from given hexadecimal representation.
98 Set the AES decryption initialization vector binary block from given hexadecimal representation.
101 Accepted URL formats:
109 Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
111 For example, to convert a GIF file given inline with @command{ffmpeg}:
113 ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
118 File access protocol.
120 Allow to read from or read to a file.
122 For example to read from a file @file{input.mpeg} with @command{ffmpeg}
125 ffmpeg -i file:input.mpeg output.mpeg
128 The ff* tools default to the file protocol, that is a resource
129 specified with the name "FILE.mpeg" is interpreted as the URL
132 This protocol accepts the following options:
136 Truncate existing files on write, if set to 1. A value of 0 prevents
137 truncating. Default value is 1.
140 Set I/O operation maximum block size, in bytes. Default value is
141 @code{INT_MAX}, which results in not limiting the requested block size.
142 Setting this value reasonably low improves user termination request reaction
143 time, which is valuable for files on slow medium.
148 FTP (File Transfer Protocol).
150 Allow to read from or write to remote resources using FTP protocol.
152 Following syntax is required.
154 ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
157 This protocol accepts the following options.
161 Set timeout of socket I/O operations used by the underlying low level
162 operation. By default it is set to -1, which means that the timeout is
165 @item ftp-anonymous-password
166 Password used when login as anonymous user. Typically an e-mail address
169 @item ftp-write-seekable
170 Control seekability of connection during encoding. If set to 1 the
171 resource is supposed to be seekable, if set to 0 it is assumed not
172 to be seekable. Default value is 0.
175 NOTE: Protocol can be used as output, but it is recommended to not do
176 it, unless special care is taken (tests, customized server configuration
177 etc.). Different FTP servers behave in different way during seek
178 operation. ff* tools may produce incomplete content due to server limitations.
186 Read Apple HTTP Live Streaming compliant segmented stream as
187 a uniform one. The M3U8 playlists describing the segments can be
188 remote HTTP resources or local files, accessed using the standard
190 The nested protocol is declared by specifying
191 "+@var{proto}" after the hls URI scheme name, where @var{proto}
192 is either "file" or "http".
195 hls+http://host/path/to/remote/resource.m3u8
196 hls+file://path/to/local/resource.m3u8
199 Using this protocol is discouraged - the hls demuxer should work
200 just as well (if not, please report the issues) and is more complete.
201 To use the hls demuxer instead, simply use the direct URLs to the
206 HTTP (Hyper Text Transfer Protocol).
208 This protocol accepts the following options.
212 Control seekability of connection. If set to 1 the resource is
213 supposed to be seekable, if set to 0 it is assumed not to be seekable,
214 if set to -1 it will try to autodetect if it is seekable. Default
218 If set to 1 use chunked transfer-encoding for posts, default is 1.
221 Set custom HTTP headers, can override built in default headers. The
222 value must be a string encoding the headers.
225 Force a content type.
228 Override User-Agent header. If not specified the protocol will use a
229 string describing the libavformat build.
231 @item multiple_requests
232 Use persistent connections if set to 1. By default it is 0.
235 Set custom HTTP post data.
238 Set timeout of socket I/O operations used by the underlying low level
239 operation. By default it is set to -1, which means that the timeout is
246 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
247 supports this, the metadata has to be retrieved by the application by reading
248 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
251 @item icy_metadata_headers
252 If the server supports ICY metadata, this contains the ICY specific HTTP reply
253 headers, separated with newline characters.
255 @item icy_metadata_packet
256 If the server supports ICY metadata, and @option{icy} was set to 1, this
257 contains the last non-empty metadata packet sent by the server.
260 Set the cookies to be sent in future requests. The format of each cookie is the
261 same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
262 delimited by a newline character.
265 @subsection HTTP Cookies
267 Some HTTP requests will be denied unless cookie values are passed in with the
268 request. The @option{cookies} option allows these cookies to be specified. At
269 the very least, each cookie must specify a value along with a path and domain.
270 HTTP requests that match both the domain and path will automatically include the
271 cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
274 The required syntax to play a stream specifying a cookie is:
276 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
281 MMS (Microsoft Media Server) protocol over TCP.
285 MMS (Microsoft Media Server) protocol over HTTP.
287 The required syntax is:
289 mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
296 Computes the MD5 hash of the data to be written, and on close writes
297 this to the designated output or stdout if none is specified. It can
298 be used to test muxers without writing an actual file.
300 Some examples follow.
302 # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
303 ffmpeg -i input.flv -f avi -y md5:output.avi.md5
305 # Write the MD5 hash of the encoded AVI file to stdout.
306 ffmpeg -i input.flv -f avi -y md5:
309 Note that some formats (typically MOV) require the output protocol to
310 be seekable, so they will fail with the MD5 output protocol.
314 UNIX pipe access protocol.
316 Allow to read and write from UNIX pipes.
318 The accepted syntax is:
323 @var{number} is the number corresponding to the file descriptor of the
324 pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
325 is not specified, by default the stdout file descriptor will be used
326 for writing, stdin for reading.
328 For example to read from stdin with @command{ffmpeg}:
330 cat test.wav | ffmpeg -i pipe:0
331 # ...this is the same as...
332 cat test.wav | ffmpeg -i pipe:
335 For writing to stdout with @command{ffmpeg}:
337 ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
338 # ...this is the same as...
339 ffmpeg -i test.wav -f avi pipe: | cat > test.avi
342 This protocol accepts the following options:
346 Set I/O operation maximum block size, in bytes. Default value is
347 @code{INT_MAX}, which results in not limiting the requested block size.
348 Setting this value reasonably low improves user termination request reaction
349 time, which is valuable if data transmission is slow.
352 Note that some formats (typically MOV), require the output protocol to
353 be seekable, so they will fail with the pipe output protocol.
357 Real-Time Messaging Protocol.
359 The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
360 content across a TCP/IP network.
362 The required syntax is:
364 rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
367 The accepted parameters are:
371 The address of the RTMP server.
374 The number of the TCP port to use (by default is 1935).
377 It is the name of the application to access. It usually corresponds to
378 the path where the application is installed on the RTMP server
379 (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
380 the value parsed from the URI through the @code{rtmp_app} option, too.
383 It is the path or name of the resource to play with reference to the
384 application specified in @var{app}, may be prefixed by "mp4:". You
385 can override the value parsed from the URI through the @code{rtmp_playpath}
389 Act as a server, listening for an incoming connection.
392 Maximum time to wait for the incoming connection. Implies listen.
395 Additionally, the following parameters can be set via command line options
396 (or in code via @code{AVOption}s):
400 Name of application to connect on the RTMP server. This option
401 overrides the parameter specified in the URI.
404 Set the client buffer time in milliseconds. The default is 3000.
407 Extra arbitrary AMF connection parameters, parsed from a string,
408 e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
409 Each value is prefixed by a single character denoting the type,
410 B for Boolean, N for number, S for string, O for object, or Z for null,
411 followed by a colon. For Booleans the data must be either 0 or 1 for
412 FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
413 1 to end or begin an object, respectively. Data items in subobjects may
414 be named, by prefixing the type with 'N' and specifying the name before
415 the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
416 times to construct arbitrary AMF sequences.
419 Version of the Flash plugin used to run the SWF player. The default
422 @item rtmp_flush_interval
423 Number of packets flushed in the same request (RTMPT only). The default
427 Specify that the media is a live stream. No resuming or seeking in
428 live streams is possible. The default value is @code{any}, which means the
429 subscriber first tries to play the live stream specified in the
430 playpath. If a live stream of that name is not found, it plays the
431 recorded stream. The other possible values are @code{live} and
435 URL of the web page in which the media was embedded. By default no
439 Stream identifier to play or to publish. This option overrides the
440 parameter specified in the URI.
443 Name of live stream to subscribe to. By default no value will be sent.
444 It is only sent if the option is specified or if rtmp_live
448 SHA256 hash of the decompressed SWF file (32 bytes).
451 Size of the decompressed SWF file, required for SWFVerification.
454 URL of the SWF player for the media. By default no value will be sent.
457 URL to player swf file, compute hash/size automatically.
460 URL of the target stream. Defaults to proto://host[:port]/app.
464 For example to read with @command{ffplay} a multimedia resource named
465 "sample" from the application "vod" from an RTMP server "myserver":
467 ffplay rtmp://myserver/vod/sample
472 Encrypted Real-Time Messaging Protocol.
474 The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
475 streaming multimedia content within standard cryptographic primitives,
476 consisting of Diffie-Hellman key exchange and HMACSHA256, generating
481 Real-Time Messaging Protocol over a secure SSL connection.
483 The Real-Time Messaging Protocol (RTMPS) is used for streaming
484 multimedia content across an encrypted connection.
488 Real-Time Messaging Protocol tunneled through HTTP.
490 The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
491 for streaming multimedia content within HTTP requests to traverse
496 Encrypted Real-Time Messaging Protocol tunneled through HTTP.
498 The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
499 is used for streaming multimedia content within HTTP requests to traverse
504 Real-Time Messaging Protocol tunneled through HTTPS.
506 The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
507 for streaming multimedia content within HTTPS requests to traverse
510 @section rtmp, rtmpe, rtmps, rtmpt, rtmpte
512 Real-Time Messaging Protocol and its variants supported through
515 Requires the presence of the librtmp headers and library during
516 configuration. You need to explicitly configure the build with
517 "--enable-librtmp". If enabled this will replace the native RTMP
520 This protocol provides most client functions and a few server
521 functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
522 encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
523 variants of these encrypted types (RTMPTE, RTMPTS).
525 The required syntax is:
527 @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
530 where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
531 "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
532 @var{server}, @var{port}, @var{app} and @var{playpath} have the same
533 meaning as specified for the RTMP native protocol.
534 @var{options} contains a list of space-separated options of the form
537 See the librtmp manual page (man 3 librtmp) for more information.
539 For example, to stream a file in real-time to an RTMP server using
542 ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
545 To play the same stream using @command{ffplay}:
547 ffplay "rtmp://myserver/live/mystream live=1"
556 RTSP is not technically a protocol handler in libavformat, it is a demuxer
557 and muxer. The demuxer supports both normal RTSP (with data transferred
558 over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
559 data transferred over RDT).
561 The muxer can be used to send a stream using RTSP ANNOUNCE to a server
562 supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
563 @uref{http://github.com/revmischa/rtsp-server, RTSP server}).
565 The required syntax for a RTSP url is:
567 rtsp://@var{hostname}[:@var{port}]/@var{path}
570 The following options (set on the @command{ffmpeg}/@command{ffplay} command
571 line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
574 Flags for @code{rtsp_transport}:
579 Use UDP as lower transport protocol.
582 Use TCP (interleaving within the RTSP control channel) as lower
586 Use UDP multicast as lower transport protocol.
589 Use HTTP tunneling as lower transport protocol, which is useful for
593 Multiple lower transport protocols may be specified, in that case they are
594 tried one at a time (if the setup of one fails, the next one is tried).
595 For the muxer, only the @code{tcp} and @code{udp} options are supported.
597 Flags for @code{rtsp_flags}:
601 Accept packets only from negotiated peer address and port.
603 Act as a server, listening for an incoming connection.
606 When receiving data over UDP, the demuxer tries to reorder received packets
607 (since they may arrive out of order, or packets may get lost totally). This
608 can be disabled by setting the maximum demuxing delay to zero (via
609 the @code{max_delay} field of AVFormatContext).
611 When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
612 streams to display can be chosen with @code{-vst} @var{n} and
613 @code{-ast} @var{n} for video and audio respectively, and can be switched
614 on the fly by pressing @code{v} and @code{a}.
616 Example command lines:
618 To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
621 ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
624 To watch a stream tunneled over HTTP:
627 ffplay -rtsp_transport http rtsp://server/video.mp4
630 To send a stream in realtime to a RTSP server, for others to watch:
633 ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
636 To receive a stream in realtime:
639 ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
644 Socket IO timeout in micro seconds.
649 Session Announcement Protocol (RFC 2974). This is not technically a
650 protocol handler in libavformat, it is a muxer and demuxer.
651 It is used for signalling of RTP streams, by announcing the SDP for the
652 streams regularly on a separate port.
656 The syntax for a SAP url given to the muxer is:
658 sap://@var{destination}[:@var{port}][?@var{options}]
661 The RTP packets are sent to @var{destination} on port @var{port},
662 or to port 5004 if no port is specified.
663 @var{options} is a @code{&}-separated list. The following options
668 @item announce_addr=@var{address}
669 Specify the destination IP address for sending the announcements to.
670 If omitted, the announcements are sent to the commonly used SAP
671 announcement multicast address 224.2.127.254 (sap.mcast.net), or
672 ff0e::2:7ffe if @var{destination} is an IPv6 address.
674 @item announce_port=@var{port}
675 Specify the port to send the announcements on, defaults to
676 9875 if not specified.
679 Specify the time to live value for the announcements and RTP packets,
682 @item same_port=@var{0|1}
683 If set to 1, send all RTP streams on the same port pair. If zero (the
684 default), all streams are sent on unique ports, with each stream on a
685 port 2 numbers higher than the previous.
686 VLC/Live555 requires this to be set to 1, to be able to receive the stream.
687 The RTP stack in libavformat for receiving requires all streams to be sent
691 Example command lines follow.
693 To broadcast a stream on the local subnet, for watching in VLC:
696 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
699 Similarly, for watching in @command{ffplay}:
702 ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
705 And for watching in @command{ffplay}, over IPv6:
708 ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
713 The syntax for a SAP url given to the demuxer is:
715 sap://[@var{address}][:@var{port}]
718 @var{address} is the multicast address to listen for announcements on,
719 if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
720 is the port that is listened on, 9875 if omitted.
722 The demuxers listens for announcements on the given address and port.
723 Once an announcement is received, it tries to receive that particular stream.
725 Example command lines follow.
727 To play back the first stream announced on the normal SAP multicast address:
733 To play back the first stream announced on one the default IPv6 SAP multicast address:
736 ffplay sap://[ff0e::2:7ffe]
741 Stream Control Transmission Protocol.
743 The accepted URL syntax is:
745 sctp://@var{host}:@var{port}[?@var{options}]
748 The protocol accepts the following options:
751 If set to any value, listen for an incoming connection. Outgoing connection is done by default.
754 Set the maximum number of streams. By default no limit is set.
759 Secure Real-time Transport Protocol.
761 The accepted options are:
765 Select input and output encoding suites.
769 @item AES_CM_128_HMAC_SHA1_80
770 @item SRTP_AES128_CM_HMAC_SHA1_80
771 @item AES_CM_128_HMAC_SHA1_32
772 @item SRTP_AES128_CM_HMAC_SHA1_32
776 @item srtp_out_params
777 Set input and output encoding parameters, which are expressed by a
778 base64-encoded representation of a binary block. The first 16 bytes of
779 this binary block are used as master key, the following 14 bytes are
785 Trasmission Control Protocol.
787 The required syntax for a TCP url is:
789 tcp://@var{hostname}:@var{port}[?@var{options}]
795 Listen for an incoming connection
797 @item timeout=@var{microseconds}
798 In read mode: if no data arrived in more than this time interval, raise error.
799 In write mode: if socket cannot be written in more than this time interval, raise error.
800 This also sets timeout on TCP connection establishing.
803 ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
804 ffplay tcp://@var{hostname}:@var{port}
811 Transport Layer Security/Secure Sockets Layer
813 The required syntax for a TLS/SSL url is:
815 tls://@var{hostname}:@var{port}[?@var{options}]
821 Act as a server, listening for an incoming connection.
823 @item cafile=@var{filename}
824 Certificate authority file. The file must be in OpenSSL PEM format.
826 @item cert=@var{filename}
827 Certificate file. The file must be in OpenSSL PEM format.
829 @item key=@var{filename}
832 @item verify=@var{0|1}
833 Verify the peer's certificate.
837 Example command lines:
839 To create a TLS/SSL server that serves an input stream.
842 ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
845 To play back a stream from the TLS/SSL server using @command{ffplay}:
848 ffplay tls://@var{hostname}:@var{port}
853 User Datagram Protocol.
855 The required syntax for a UDP url is:
857 udp://@var{hostname}:@var{port}[?@var{options}]
860 @var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
862 In case threading is enabled on the system, a circular buffer is used
863 to store the incoming data, which allows to reduce loss of data due to
864 UDP socket buffer overruns. The @var{fifo_size} and
865 @var{overrun_nonfatal} options are related to this buffer.
867 The list of supported options follows.
871 @item buffer_size=@var{size}
872 Set the UDP socket buffer size in bytes. This is used both for the
873 receiving and the sending buffer size.
875 @item localport=@var{port}
876 Override the local UDP port to bind with.
878 @item localaddr=@var{addr}
879 Choose the local IP address. This is useful e.g. if sending multicast
880 and the host has multiple interfaces, where the user can choose
881 which interface to send on by specifying the IP address of that interface.
883 @item pkt_size=@var{size}
884 Set the size in bytes of UDP packets.
886 @item reuse=@var{1|0}
887 Explicitly allow or disallow reusing UDP sockets.
890 Set the time to live value (for multicast only).
892 @item connect=@var{1|0}
893 Initialize the UDP socket with @code{connect()}. In this case, the
894 destination address can't be changed with ff_udp_set_remote_url later.
895 If the destination address isn't known at the start, this option can
896 be specified in ff_udp_set_remote_url, too.
897 This allows finding out the source address for the packets with getsockname,
898 and makes writes return with AVERROR(ECONNREFUSED) if "destination
899 unreachable" is received.
900 For receiving, this gives the benefit of only receiving packets from
901 the specified peer address/port.
903 @item sources=@var{address}[,@var{address}]
904 Only receive packets sent to the multicast group from one of the
905 specified sender IP addresses.
907 @item block=@var{address}[,@var{address}]
908 Ignore packets sent to the multicast group from the specified
911 @item fifo_size=@var{units}
912 Set the UDP receiving circular buffer size, expressed as a number of
913 packets with size of 188 bytes. If not specified defaults to 7*4096.
915 @item overrun_nonfatal=@var{1|0}
916 Survive in case of UDP receiving circular buffer overrun. Default
919 @item timeout=@var{microseconds}
920 In read mode: if no data arrived in more than this time interval, raise error.
923 Some usage examples of the UDP protocol with @command{ffmpeg} follow.
925 To stream over UDP to a remote endpoint:
927 ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
930 To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
932 ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
935 To receive over UDP from a remote endpoint:
937 ffmpeg -i udp://[@var{multicast-address}]:@var{port}
944 The required syntax for a Unix socket URL is:
947 unix://@var{filepath}
950 The following parameters can be set via command line options
951 (or in code via @code{AVOption}s):
957 Create the Unix socket in listening mode.