protocol.
For example to read a sequence of files @file{split1.mpeg},
-@file{split2.mpeg}, @file{split3.mpeg} with @file{avplay} use the
+@file{split2.mpeg}, @file{split3.mpeg} with @command{avplay} use the
command:
@example
avplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
hls+file://path/to/local/resource.m3u8
@end example
+Using this protocol is discouraged - the hls demuxer should work
+just as well (if not, please report the issues) and is more complete.
+To use the hls demuxer instead, simply use the direct URLs to the
+m3u8 files.
+
@section http
HTTP (Hyper Text Transfer Protocol).
The required syntax is:
@example
-rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
+rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
@end example
The accepted parameters are:
@item app
It is the name of the application to access. It usually corresponds to
the path where the application is installed on the RTMP server
-(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.).
+(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
+the value parsed from the URI through the @code{rtmp_app} option, too.
@item playpath
It is the path or name of the resource to play with reference to the
-application specified in @var{app}, may be prefixed by "mp4:".
+application specified in @var{app}, may be prefixed by "mp4:". You
+can override the value parsed from the URI through the @code{rtmp_playpath}
+option, too.
+
+@item listen
+Act as a server, listening for an incoming connection.
+
+@item timeout
+Maximum time to wait for the incoming connection. Implies listen.
+@end table
+
+Additionally, the following parameters can be set via command line options
+(or in code via @code{AVOption}s):
+@table @option
+
+@item rtmp_app
+Name of application to connect on the RTMP server. This option
+overrides the parameter specified in the URI.
+
+@item rtmp_buffer
+Set the client buffer time in milliseconds. The default is 3000.
+
+@item rtmp_conn
+Extra arbitrary AMF connection parameters, parsed from a string,
+e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
+Each value is prefixed by a single character denoting the type,
+B for Boolean, N for number, S for string, O for object, or Z for null,
+followed by a colon. For Booleans the data must be either 0 or 1 for
+FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
+1 to end or begin an object, respectively. Data items in subobjects may
+be named, by prefixing the type with 'N' and specifying the name before
+the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
+times to construct arbitrary AMF sequences.
+
+@item rtmp_flashver
+Version of the Flash plugin used to run the SWF player. The default
+is LNX 9,0,124,2.
+
+@item rtmp_flush_interval
+Number of packets flushed in the same request (RTMPT only). The default
+is 10.
+
+@item rtmp_live
+Specify that the media is a live stream. No resuming or seeking in
+live streams is possible. The default value is @code{any}, which means the
+subscriber first tries to play the live stream specified in the
+playpath. If a live stream of that name is not found, it plays the
+recorded stream. The other possible values are @code{live} and
+@code{recorded}.
+
+@item rtmp_pageurl
+URL of the web page in which the media was embedded. By default no
+value will be sent.
+
+@item rtmp_playpath
+Stream identifier to play or to publish. This option overrides the
+parameter specified in the URI.
+
+@item rtmp_subscribe
+Name of live stream to subscribe to. By default no value will be sent.
+It is only sent if the option is specified or if rtmp_live
+is set to live.
+
+@item rtmp_swfhash
+SHA256 hash of the decompressed SWF file (32 bytes).
+
+@item rtmp_swfsize
+Size of the decompressed SWF file, required for SWFVerification.
+
+@item rtmp_swfurl
+URL of the SWF player for the media. By default no value will be sent.
+
+@item rtmp_swfverify
+URL to player swf file, compute hash/size automatically.
+
+@item rtmp_tcurl
+URL of the target stream. Defaults to proto://host[:port]/app.
@end table
-For example to read with @file{avplay} a multimedia resource named
+For example to read with @command{avplay} a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
@example
avplay rtmp://myserver/vod/sample
@end example
+@section rtmpe
+
+Encrypted Real-Time Messaging Protocol.
+
+The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
+streaming multimedia content within standard cryptographic primitives,
+consisting of Diffie-Hellman key exchange and HMACSHA256, generating
+a pair of RC4 keys.
+
+@section rtmps
+
+Real-Time Messaging Protocol over a secure SSL connection.
+
+The Real-Time Messaging Protocol (RTMPS) is used for streaming
+multimedia content across an encrypted connection.
+
+@section rtmpt
+
+Real-Time Messaging Protocol tunneled through HTTP.
+
+The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
+for streaming multimedia content within HTTP requests to traverse
+firewalls.
+
+@section rtmpte
+
+Encrypted Real-Time Messaging Protocol tunneled through HTTP.
+
+The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
+is used for streaming multimedia content within HTTP requests to traverse
+firewalls.
+
+@section rtmpts
+
+Real-Time Messaging Protocol tunneled through HTTPS.
+
+The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
+for streaming multimedia content within HTTPS requests to traverse
+firewalls.
+
@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through
avconv -re -i myfile -f flv rtmp://myserver/live/mystream
@end example
-To play the same stream using @file{avplay}:
+To play the same stream using @command{avplay}:
@example
avplay "rtmp://myserver/live/mystream live=1"
@end example
rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example
-The following options (set on the @command{avconv}/@file{avplay} command
+The following options (set on the @command{avconv}/@command{avplay} command
line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
are supported:
@table @option
@item filter_src
Accept packets only from negotiated peer address and port.
+@item listen
+Act as a server, listening for an incoming connection.
@end table
When receiving data over UDP, the demuxer tries to reorder received packets
-(since they may arrive out of order, or packets may get lost totally). In
-order for this to be enabled, a maximum delay must be specified in the
-@code{max_delay} field of AVFormatContext.
+(since they may arrive out of order, or packets may get lost totally). This
+can be disabled by setting the maximum demuxing delay to zero (via
+the @code{max_delay} field of AVFormatContext).
-When watching multi-bitrate Real-RTSP streams with @file{avplay}, the
+When watching multi-bitrate Real-RTSP streams with @command{avplay}, the
streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
on the fly by pressing @code{v} and @code{a}.
avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example
+To receive a stream in realtime:
+
+@example
+avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
+@end example
+
@section sap
Session Announcement Protocol (RFC 2974). This is not technically a
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.
projects / ffmpeg / blobdiff