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This way is possible to have simple micro-server act like ffmpeg -i file.nut -vcodec copy -acodec copy -f nut tcp://foo:1234?listen
481 lines
13 KiB
Plaintext
481 lines
13 KiB
Plaintext
@chapter Protocols
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@c man begin PROTOCOLS
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Protocols are configured elements in Libav which allow to access
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resources which require the use of a particular protocol.
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When you configure your Libav build, all the supported protocols are
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enabled by default. You can list all available ones using the
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configure option "--list-protocols".
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You can disable all the protocols using the configure option
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"--disable-protocols", and selectively enable a protocol using the
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option "--enable-protocol=@var{PROTOCOL}", or you can disable a
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particular protocol using the option
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"--disable-protocol=@var{PROTOCOL}".
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The option "-protocols" of the ff* tools will display the list of
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supported protocols.
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A description of the currently available protocols follows.
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@section applehttp
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Read Apple HTTP Live Streaming compliant segmented stream as
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a uniform one. The M3U8 playlists describing the segments can be
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remote HTTP resources or local files, accessed using the standard
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file protocol.
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HTTP is default, specific protocol can be declared by specifying
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"+@var{proto}" after the applehttp URI scheme name, where @var{proto}
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is either "file" or "http".
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@example
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applehttp://host/path/to/remote/resource.m3u8
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applehttp+http://host/path/to/remote/resource.m3u8
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applehttp+file://path/to/local/resource.m3u8
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@end example
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@section concat
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Physical concatenation protocol.
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Allow to read and seek from many resource in sequence as if they were
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a unique resource.
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A URL accepted by this protocol has the syntax:
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@example
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concat:@var{URL1}|@var{URL2}|...|@var{URLN}
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@end example
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where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
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resource to be concatenated, each one possibly specifying a distinct
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protocol.
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For example to read a sequence of files @file{split1.mpeg},
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@file{split2.mpeg}, @file{split3.mpeg} with @file{ffplay} use the
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command:
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@example
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ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
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@end example
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Note that you may need to escape the character "|" which is special for
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many shells.
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@section file
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File access protocol.
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Allow to read from or read to a file.
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For example to read from a file @file{input.mpeg} with @file{ffmpeg}
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use the command:
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@example
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ffmpeg -i file:input.mpeg output.mpeg
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@end example
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The ff* tools default to the file protocol, that is a resource
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specified with the name "FILE.mpeg" is interpreted as the URL
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"file:FILE.mpeg".
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@section gopher
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Gopher protocol.
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@section http
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HTTP (Hyper Text Transfer Protocol).
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@section mmst
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MMS (Microsoft Media Server) protocol over TCP.
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@section mmsh
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MMS (Microsoft Media Server) protocol over HTTP.
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The required syntax is:
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@example
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mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
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@end example
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@section md5
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MD5 output protocol.
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Computes the MD5 hash of the data to be written, and on close writes
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this to the designated output or stdout if none is specified. It can
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be used to test muxers without writing an actual file.
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Some examples follow.
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@example
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# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
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ffmpeg -i input.flv -f avi -y md5:output.avi.md5
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# Write the MD5 hash of the encoded AVI file to stdout.
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ffmpeg -i input.flv -f avi -y md5:
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@end example
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Note that some formats (typically MOV) require the output protocol to
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be seekable, so they will fail with the MD5 output protocol.
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@section pipe
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UNIX pipe access protocol.
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Allow to read and write from UNIX pipes.
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The accepted syntax is:
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@example
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pipe:[@var{number}]
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@end example
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@var{number} is the number corresponding to the file descriptor of the
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pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
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is not specified, by default the stdout file descriptor will be used
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for writing, stdin for reading.
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For example to read from stdin with @file{ffmpeg}:
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@example
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cat test.wav | ffmpeg -i pipe:0
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# ...this is the same as...
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cat test.wav | ffmpeg -i pipe:
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@end example
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For writing to stdout with @file{ffmpeg}:
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@example
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ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
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# ...this is the same as...
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ffmpeg -i test.wav -f avi pipe: | cat > test.avi
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@end example
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Note that some formats (typically MOV), require the output protocol to
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be seekable, so they will fail with the pipe output protocol.
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@section rtmp
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Real-Time Messaging Protocol.
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The Real-Time Messaging Protocol (RTMP) is used for streaming multime‐
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dia content across a TCP/IP network.
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The required syntax is:
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@example
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rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
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@end example
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The accepted parameters are:
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@table @option
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@item server
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The address of the RTMP server.
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@item port
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The number of the TCP port to use (by default is 1935).
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@item app
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It is the name of the application to access. It usually corresponds to
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the path where the application is installed on the RTMP server
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(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.).
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@item playpath
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It is the path or name of the resource to play with reference to the
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application specified in @var{app}, may be prefixed by "mp4:".
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@end table
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For example to read with @file{ffplay} a multimedia resource named
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"sample" from the application "vod" from an RTMP server "myserver":
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@example
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ffplay rtmp://myserver/vod/sample
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@end example
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@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
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Real-Time Messaging Protocol and its variants supported through
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librtmp.
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Requires the presence of the librtmp headers and library during
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configuration. You need to explicitely configure the build with
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"--enable-librtmp". If enabled this will replace the native RTMP
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protocol.
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This protocol provides most client functions and a few server
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functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
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encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
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variants of these encrypted types (RTMPTE, RTMPTS).
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The required syntax is:
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@example
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@var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
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@end example
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where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
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"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
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@var{server}, @var{port}, @var{app} and @var{playpath} have the same
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meaning as specified for the RTMP native protocol.
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@var{options} contains a list of space-separated options of the form
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@var{key}=@var{val}.
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See the librtmp manual page (man 3 librtmp) for more information.
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For example, to stream a file in real-time to an RTMP server using
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@file{ffmpeg}:
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@example
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ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
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@end example
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To play the same stream using @file{ffplay}:
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@example
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ffplay "rtmp://myserver/live/mystream live=1"
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@end example
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@section rtp
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Real-Time Protocol.
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@section rtsp
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RTSP is not technically a protocol handler in libavformat, it is a demuxer
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and muxer. The demuxer supports both normal RTSP (with data transferred
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over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
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data transferred over RDT).
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The muxer can be used to send a stream using RTSP ANNOUNCE to a server
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supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
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RTSP server, @url{http://github.com/revmischa/rtsp-server}).
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The required syntax for a RTSP url is:
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@example
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rtsp://@var{hostname}[:@var{port}]/@var{path}[?@var{options}]
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@end example
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@var{options} is a @code{&}-separated list. The following options
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are supported:
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@table @option
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@item udp
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Use UDP as lower transport protocol.
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@item tcp
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Use TCP (interleaving within the RTSP control channel) as lower
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transport protocol.
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@item multicast
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Use UDP multicast as lower transport protocol.
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@item http
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Use HTTP tunneling as lower transport protocol, which is useful for
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passing proxies.
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@item filter_src
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Accept packets only from negotiated peer address and port.
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@end table
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Multiple lower transport protocols may be specified, in that case they are
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tried one at a time (if the setup of one fails, the next one is tried).
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For the muxer, only the @code{tcp} and @code{udp} options are supported.
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When receiving data over UDP, the demuxer tries to reorder received packets
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(since they may arrive out of order, or packets may get lost totally). In
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order for this to be enabled, a maximum delay must be specified in the
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@code{max_delay} field of AVFormatContext.
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When watching multi-bitrate Real-RTSP streams with @file{ffplay}, the
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streams to display can be chosen with @code{-vst} @var{n} and
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@code{-ast} @var{n} for video and audio respectively, and can be switched
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on the fly by pressing @code{v} and @code{a}.
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Example command lines:
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To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
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@example
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ffplay -max_delay 500000 rtsp://server/video.mp4?udp
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@end example
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To watch a stream tunneled over HTTP:
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@example
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ffplay rtsp://server/video.mp4?http
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@end example
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To send a stream in realtime to a RTSP server, for others to watch:
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@example
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ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
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@end example
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@section sap
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Session Announcement Protocol (RFC 2974). This is not technically a
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protocol handler in libavformat, it is a muxer and demuxer.
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It is used for signalling of RTP streams, by announcing the SDP for the
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streams regularly on a separate port.
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@subsection Muxer
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The syntax for a SAP url given to the muxer is:
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@example
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sap://@var{destination}[:@var{port}][?@var{options}]
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@end example
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The RTP packets are sent to @var{destination} on port @var{port},
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or to port 5004 if no port is specified.
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@var{options} is a @code{&}-separated list. The following options
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are supported:
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@table @option
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@item announce_addr=@var{address}
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Specify the destination IP address for sending the announcements to.
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If omitted, the announcements are sent to the commonly used SAP
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announcement multicast address 224.2.127.254 (sap.mcast.net), or
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ff0e::2:7ffe if @var{destination} is an IPv6 address.
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@item announce_port=@var{port}
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Specify the port to send the announcements on, defaults to
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9875 if not specified.
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@item ttl=@var{ttl}
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Specify the time to live value for the announcements and RTP packets,
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defaults to 255.
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@item same_port=@var{0|1}
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If set to 1, send all RTP streams on the same port pair. If zero (the
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default), all streams are sent on unique ports, with each stream on a
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port 2 numbers higher than the previous.
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VLC/Live555 requires this to be set to 1, to be able to receive the stream.
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The RTP stack in libavformat for receiving requires all streams to be sent
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on unique ports.
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@end table
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Example command lines follow.
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To broadcast a stream on the local subnet, for watching in VLC:
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@example
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ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
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@end example
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Similarly, for watching in ffplay:
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@example
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ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
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@end example
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And for watching in ffplay, over IPv6:
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@example
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ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
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@end example
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@subsection Demuxer
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The syntax for a SAP url given to the demuxer is:
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@example
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sap://[@var{address}][:@var{port}]
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@end example
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@var{address} is the multicast address to listen for announcements on,
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if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
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is the port that is listened on, 9875 if omitted.
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The demuxers listens for announcements on the given address and port.
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Once an announcement is received, it tries to receive that particular stream.
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Example command lines follow.
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To play back the first stream announced on the normal SAP multicast address:
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@example
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ffplay sap://
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@end example
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To play back the first stream announced on one the default IPv6 SAP multicast address:
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@example
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ffplay sap://[ff0e::2:7ffe]
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@end example
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@section tcp
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Trasmission Control Protocol.
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The required syntax for a TCP url is:
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@example
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tcp://@var{hostname}:@var{port}[?@var{options}]
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@end example
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@table @option
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@item listen
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Listen for an incoming connection
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@example
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ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
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ffplay tcp://@var{hostname}:@var{port}
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@end example
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@end table
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@section udp
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User Datagram Protocol.
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The required syntax for a UDP url is:
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@example
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udp://@var{hostname}:@var{port}[?@var{options}]
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@end example
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@var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
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Follow the list of supported options.
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@table @option
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@item buffer_size=@var{size}
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set the UDP buffer size in bytes
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@item localport=@var{port}
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override the local UDP port to bind with
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@item pkt_size=@var{size}
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set the size in bytes of UDP packets
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@item reuse=@var{1|0}
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explicitly allow or disallow reusing UDP sockets
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@item ttl=@var{ttl}
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set the time to live value (for multicast only)
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@item connect=@var{1|0}
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Initialize the UDP socket with @code{connect()}. In this case, the
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destination address can't be changed with ff_udp_set_remote_url later.
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If the destination address isn't known at the start, this option can
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be specified in ff_udp_set_remote_url, too.
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This allows finding out the source address for the packets with getsockname,
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and makes writes return with AVERROR(ECONNREFUSED) if "destination
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unreachable" is received.
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For receiving, this gives the benefit of only receiving packets from
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the specified peer address/port.
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@end table
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Some usage examples of the udp protocol with @file{ffmpeg} follow.
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To stream over UDP to a remote endpoint:
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@example
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ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
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@end example
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To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
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@example
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ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
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@end example
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To receive over UDP from a remote endpoint:
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@example
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ffmpeg -i udp://[@var{multicast-address}]:@var{port}
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@end example
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@c man end PROTOCOLS
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