630 lines
18 KiB
Plaintext
630 lines
18 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 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 @command{avplay} use the
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command:
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@example
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avplay 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 @command{avconv}
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use the command:
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@example
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avconv -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 hls
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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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The nested protocol is declared by specifying
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"+@var{proto}" after the hls URI scheme name, where @var{proto}
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is either "file" or "http".
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@example
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hls+http://host/path/to/remote/resource.m3u8
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hls+file://path/to/local/resource.m3u8
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@end example
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Using this protocol is discouraged - the hls demuxer should work
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just as well (if not, please report the issues) and is more complete.
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To use the hls demuxer instead, simply use the direct URLs to the
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m3u8 files.
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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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avconv -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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avconv -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 @command{avconv}:
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@example
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cat test.wav | avconv -i pipe:0
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# ...this is the same as...
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cat test.wav | avconv -i pipe:
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@end example
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For writing to stdout with @command{avconv}:
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@example
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avconv -i test.wav -f avi pipe:1 | cat > test.avi
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# ...this is the same as...
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avconv -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 multimedia
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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{instance}][/@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.). You can override
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the value parsed from the URI through the @code{rtmp_app} option, too.
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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:". You
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can override the value parsed from the URI through the @code{rtmp_playpath}
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option, too.
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@item listen
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Act as a server, listening for an incoming connection.
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@item timeout
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Maximum time to wait for the incoming connection. Implies listen.
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@end table
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Additionally, the following parameters can be set via command line options
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(or in code via @code{AVOption}s):
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@table @option
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@item rtmp_app
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Name of application to connect on the RTMP server. This option
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overrides the parameter specified in the URI.
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@item rtmp_buffer
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Set the client buffer time in milliseconds. The default is 3000.
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@item rtmp_conn
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Extra arbitrary AMF connection parameters, parsed from a string,
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e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
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Each value is prefixed by a single character denoting the type,
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B for Boolean, N for number, S for string, O for object, or Z for null,
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followed by a colon. For Booleans the data must be either 0 or 1 for
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FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
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1 to end or begin an object, respectively. Data items in subobjects may
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be named, by prefixing the type with 'N' and specifying the name before
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the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
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times to construct arbitrary AMF sequences.
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@item rtmp_flashver
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Version of the Flash plugin used to run the SWF player. The default
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is LNX 9,0,124,2.
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@item rtmp_flush_interval
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Number of packets flushed in the same request (RTMPT only). The default
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is 10.
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@item rtmp_live
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Specify that the media is a live stream. No resuming or seeking in
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live streams is possible. The default value is @code{any}, which means the
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subscriber first tries to play the live stream specified in the
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playpath. If a live stream of that name is not found, it plays the
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recorded stream. The other possible values are @code{live} and
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@code{recorded}.
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@item rtmp_pageurl
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URL of the web page in which the media was embedded. By default no
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value will be sent.
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@item rtmp_playpath
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Stream identifier to play or to publish. This option overrides the
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parameter specified in the URI.
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@item rtmp_subscribe
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Name of live stream to subscribe to. By default no value will be sent.
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It is only sent if the option is specified or if rtmp_live
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is set to live.
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@item rtmp_swfhash
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SHA256 hash of the decompressed SWF file (32 bytes).
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@item rtmp_swfsize
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Size of the decompressed SWF file, required for SWFVerification.
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@item rtmp_swfurl
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URL of the SWF player for the media. By default no value will be sent.
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@item rtmp_swfverify
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URL to player swf file, compute hash/size automatically.
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@item rtmp_tcurl
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URL of the target stream. Defaults to proto://host[:port]/app.
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@end table
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For example to read with @command{avplay} 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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avplay rtmp://myserver/vod/sample
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@end example
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@section rtmpe
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Encrypted Real-Time Messaging Protocol.
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The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
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streaming multimedia content within standard cryptographic primitives,
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consisting of Diffie-Hellman key exchange and HMACSHA256, generating
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a pair of RC4 keys.
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@section rtmps
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Real-Time Messaging Protocol over a secure SSL connection.
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The Real-Time Messaging Protocol (RTMPS) is used for streaming
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multimedia content across an encrypted connection.
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@section rtmpt
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Real-Time Messaging Protocol tunneled through HTTP.
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The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
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for streaming multimedia content within HTTP requests to traverse
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firewalls.
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@section rtmpte
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Encrypted Real-Time Messaging Protocol tunneled through HTTP.
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The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
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is used for streaming multimedia content within HTTP requests to traverse
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firewalls.
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@section rtmpts
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Real-Time Messaging Protocol tunneled through HTTPS.
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The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
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for streaming multimedia content within HTTPS requests to traverse
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firewalls.
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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 explicitly 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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@command{avconv}:
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@example
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avconv -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 @command{avplay}:
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@example
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avplay "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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@uref{http://github.com/revmischa/rtsp-server, 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}
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@end example
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The following options (set on the @command{avconv}/@command{avplay} command
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line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
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are supported:
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Flags for @code{rtsp_transport}:
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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 udp_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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@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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Flags for @code{rtsp_flags}:
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@table @option
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@item filter_src
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Accept packets only from negotiated peer address and port.
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@item listen
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Act as a server, listening for an incoming connection.
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@end table
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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). This
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can be disabled by setting the maximum demuxing delay to zero (via
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the @code{max_delay} field of AVFormatContext).
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When watching multi-bitrate Real-RTSP streams with @command{avplay}, 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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avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
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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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avplay -rtsp_transport http rtsp://server/video.mp4
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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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avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
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@end example
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To receive a stream in realtime:
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@example
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avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
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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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avconv -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 avplay:
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@example
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avconv -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 avplay, over IPv6:
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@example
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avconv -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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avplay 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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avplay sap://[ff0e::2:7ffe]
|
|
@end example
|
|
|
|
@section tcp
|
|
|
|
Trasmission Control Protocol.
|
|
|
|
The required syntax for a TCP url is:
|
|
@example
|
|
tcp://@var{hostname}:@var{port}[?@var{options}]
|
|
@end example
|
|
|
|
@table @option
|
|
|
|
@item listen
|
|
Listen for an incoming connection
|
|
|
|
@example
|
|
avconv -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
|
|
avplay tcp://@var{hostname}:@var{port}
|
|
@end example
|
|
|
|
@end table
|
|
|
|
@section udp
|
|
|
|
User Datagram Protocol.
|
|
|
|
The required syntax for a UDP url is:
|
|
@example
|
|
udp://@var{hostname}:@var{port}[?@var{options}]
|
|
@end example
|
|
|
|
@var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
|
|
Follow the list of supported options.
|
|
|
|
@table @option
|
|
|
|
@item buffer_size=@var{size}
|
|
set the UDP buffer size in bytes
|
|
|
|
@item localport=@var{port}
|
|
override the local UDP port to bind with
|
|
|
|
@item localaddr=@var{addr}
|
|
Choose the local IP address. This is useful e.g. if sending multicast
|
|
and the host has multiple interfaces, where the user can choose
|
|
which interface to send on by specifying the IP address of that interface.
|
|
|
|
@item pkt_size=@var{size}
|
|
set the size in bytes of UDP packets
|
|
|
|
@item reuse=@var{1|0}
|
|
explicitly allow or disallow reusing UDP sockets
|
|
|
|
@item ttl=@var{ttl}
|
|
set the time to live value (for multicast only)
|
|
|
|
@item connect=@var{1|0}
|
|
Initialize the UDP socket with @code{connect()}. In this case, the
|
|
destination address can't be changed with ff_udp_set_remote_url later.
|
|
If the destination address isn't known at the start, this option can
|
|
be specified in ff_udp_set_remote_url, too.
|
|
This allows finding out the source address for the packets with getsockname,
|
|
and makes writes return with AVERROR(ECONNREFUSED) if "destination
|
|
unreachable" is received.
|
|
For receiving, this gives the benefit of only receiving packets from
|
|
the specified peer address/port.
|
|
|
|
@item sources=@var{address}[,@var{address}]
|
|
Only receive packets sent to the multicast group from one of the
|
|
specified sender IP addresses.
|
|
|
|
@item block=@var{address}[,@var{address}]
|
|
Ignore packets sent to the multicast group from the specified
|
|
sender IP addresses.
|
|
@end table
|
|
|
|
Some usage examples of the udp protocol with @command{avconv} follow.
|
|
|
|
To stream over UDP to a remote endpoint:
|
|
@example
|
|
avconv -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
|
|
@end example
|
|
|
|
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
|
|
@example
|
|
avconv -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
|
|
@end example
|
|
|
|
To receive over UDP from a remote endpoint:
|
|
@example
|
|
avconv -i udp://[@var{multicast-address}]:@var{port}
|
|
@end example
|
|
|
|
@c man end PROTOCOLS
|