@chapter Protocols
@c man begin PROTOCOLS
-Protocols are configured elements in FFmpeg which allow to access
+Protocols are configured elements in Libav which allow to access
resources which require the use of a particular protocol.
-When you configure your FFmpeg build, all the supported protocols are
+When you configure your Libav build, all the supported protocols are
enabled by default. You can list all available ones using the
configure option "--list-protocols".
protocol.
For example to read a sequence of files @file{split1.mpeg},
-@file{split2.mpeg}, @file{split3.mpeg} with @file{ffplay} use the
+@file{split2.mpeg}, @file{split3.mpeg} with @command{avplay} use the
command:
@example
-ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
+avplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
@end example
Note that you may need to escape the character "|" which is special for
Allow to read from or read to a file.
-For example to read from a file @file{input.mpeg} with @file{ffmpeg}
+For example to read from a file @file{input.mpeg} with @command{avconv}
use the command:
@example
-ffmpeg -i file:input.mpeg output.mpeg
+avconv -i file:input.mpeg output.mpeg
@end example
The ff* tools default to the file protocol, that is a resource
Gopher protocol.
+@section hls
+
+Read Apple HTTP Live Streaming compliant segmented stream as
+a uniform one. The M3U8 playlists describing the segments can be
+remote HTTP resources or local files, accessed using the standard
+file protocol.
+The nested protocol is declared by specifying
+"+@var{proto}" after the hls URI scheme name, where @var{proto}
+is either "file" or "http".
+
+@example
+hls+http://host/path/to/remote/resource.m3u8
+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).
Some examples follow.
@example
# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
-ffmpeg -i input.flv -f avi -y md5:output.avi.md5
+avconv -i input.flv -f avi -y md5:output.avi.md5
# Write the MD5 hash of the encoded AVI file to stdout.
-ffmpeg -i input.flv -f avi -y md5:
+avconv -i input.flv -f avi -y md5:
@end example
Note that some formats (typically MOV) require the output protocol to
is not specified, by default the stdout file descriptor will be used
for writing, stdin for reading.
-For example to read from stdin with @file{ffmpeg}:
+For example to read from stdin with @command{avconv}:
@example
-cat test.wav | ffmpeg -i pipe:0
+cat test.wav | avconv -i pipe:0
# ...this is the same as...
-cat test.wav | ffmpeg -i pipe:
+cat test.wav | avconv -i pipe:
@end example
-For writing to stdout with @file{ffmpeg}:
+For writing to stdout with @command{avconv}:
@example
-ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
+avconv -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
-ffmpeg -i test.wav -f avi pipe: | cat > test.avi
+avconv -i test.wav -f avi pipe: | cat > test.avi
@end example
Note that some formats (typically MOV), require the output protocol to
Real-Time Messaging Protocol.
-The Real-Time Messaging Protocol (RTMP) is used for streaming multime‐
-dia content across a TCP/IP network.
+The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
+content across a TCP/IP network.
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
-For example to read with @file{ffplay} a multimedia resource named
+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 @command{avplay} a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
@example
-ffplay rtmp://myserver/vod/sample
+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
librtmp.
Requires the presence of the librtmp headers and library during
-configuration. You need to explicitely configure the build with
+configuration. You need to explicitly configure the build with
"--enable-librtmp". If enabled this will replace the native RTMP
protocol.
See the librtmp manual page (man 3 librtmp) for more information.
For example, to stream a file in real-time to an RTMP server using
-@file{ffmpeg}:
+@command{avconv}:
@example
-ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
+avconv -re -i myfile -f flv rtmp://myserver/live/mystream
@end example
-To play the same stream using @file{ffplay}:
+To play the same stream using @command{avplay}:
@example
-ffplay "rtmp://myserver/live/mystream live=1"
+avplay "rtmp://myserver/live/mystream live=1"
@end example
@section rtp
The muxer can be used to send a stream using RTSP ANNOUNCE to a server
supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
-RTSP server, @url{http://github.com/revmischa/rtsp-server}).
+@uref{http://github.com/revmischa/rtsp-server, RTSP server}).
The required syntax for a RTSP url is:
@example
-rtsp://@var{hostname}[:@var{port}]/@var{path}[?@var{options}]
+rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example
-@var{options} is a @code{&}-separated list. The following options
+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:
+Flags for @code{rtsp_transport}:
+
@table @option
@item udp
Use TCP (interleaving within the RTSP control channel) as lower
transport protocol.
-@item multicast
+@item udp_multicast
Use UDP multicast as lower transport protocol.
@item http
Use HTTP tunneling as lower transport protocol, which is useful for
passing proxies.
-
-@item filter_src
-Accept packets only from negotiated peer address and port.
@end table
Multiple lower transport protocols may be specified, in that case they are
tried one at a time (if the setup of one fails, the next one is tried).
For the muxer, only the @code{tcp} and @code{udp} options are supported.
+Flags for @code{rtsp_flags}:
+
+@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{ffplay}, 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}.
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
@example
-ffplay -max_delay 500000 rtsp://server/video.mp4?udp
+avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
@end example
To watch a stream tunneled over HTTP:
@example
-ffplay rtsp://server/video.mp4?http
+avplay -rtsp_transport http rtsp://server/video.mp4
@end example
To send a stream in realtime to a RTSP server, for others to watch:
@example
-ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
+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
To broadcast a stream on the local subnet, for watching in VLC:
@example
-ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
+avconv -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
@end example
-Similarly, for watching in ffplay:
+Similarly, for watching in avplay:
@example
-ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
+avconv -re -i @var{input} -f sap sap://224.0.0.255
@end example
-And for watching in ffplay, over IPv6:
+And for watching in avplay, over IPv6:
@example
-ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
+avconv -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
@end example
@subsection Demuxer
To play back the first stream announced on the normal SAP multicast address:
@example
-ffplay sap://
+avplay sap://
@end example
To play back the first stream announced on one the default IPv6 SAP multicast address:
@example
-ffplay sap://[ff0e::2:7ffe]
+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.
@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 connect=@var{1|0}
Initialize the UDP socket with @code{connect()}. In this case, the
-destination address can't be changed with udp_set_remote_url later.
+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 udp_set_remote_url, too.
+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 @file{ffmpeg} follow.
+Some usage examples of the udp protocol with @command{avconv} follow.
To stream over UDP to a remote endpoint:
@example
-ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
+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
-ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
+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
-ffmpeg -i udp://[@var{multicast-address}]:@var{port}
+avconv -i udp://[@var{multicast-address}]:@var{port}
@end example
@c man end PROTOCOLS
projects / ffmpeg / blobdiff