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    @chapter Protocols
    @c man begin PROTOCOLS
    
    Protocols are configured elements in FFmpeg which allow to access
    resources which require the use of a particular protocol.
    
    
    When you configure your FFmpeg build, all the supported protocols are
    enabled by default. You can list all available ones using the
    configure option "--list-protocols".
    
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    You can disable all the protocols using the configure option
    "--disable-protocols", and selectively enable a protocol using the
    option "--enable-protocol=@var{PROTOCOL}", or you can disable a
    particular protocol using the option
    "--disable-protocol=@var{PROTOCOL}".
    
    The option "-protocols" of the ff* tools will display the list of
    
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    A description of the currently available protocols follows.
    
    @section concat
    
    Physical concatenation protocol.
    
    
    Allow to read and seek from many resource in sequence as if they were
    a unique resource.
    
    A URL accepted by this protocol has the syntax:
    
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    @example
    concat:@var{URL1}|@var{URL2}|...|@var{URLN}
    @end example
    
    where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
    resource to be concatenated, each one possibly specifying a distinct
    protocol.
    
    For example to read a sequence of files @file{split1.mpeg},
    @file{split2.mpeg}, @file{split3.mpeg} with @file{ffplay} use the
    command:
    @example
    ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
    @end example
    
    Note that you may need to escape the character "|" which is special for
    many shells.
    
    @section file
    
    File access protocol.
    
    Allow to read from or read to a file.
    
    For example to read from a file @file{input.mpeg} with @file{ffmpeg}
    use the command:
    @example
    ffmpeg -i file:input.mpeg output.mpeg
    @end example
    
    
    The ff* tools default to the file protocol, that is a resource
    specified with the name "FILE.mpeg" is interpreted as the URL
    "file:FILE.mpeg".
    
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    @section gopher
    
    Gopher protocol.
    
    @section http
    
    
    HTTP (Hyper Text Transfer Protocol).
    
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    @section mmst
    
    MMS (Microsoft Media Server) protocol over TCP.
    
    
    @section mmsh
    
    MMS (Microsoft Media Server) protocol over HTTP.
    
    The required syntax is:
    @example
    mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
    @end example
    
    
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    @section md5
    
    MD5 output protocol.
    
    
    Computes the MD5 hash of the data to be written, and on close writes
    this to the designated output or stdout if none is specified. It can
    be used to test muxers without writing an actual file.
    
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    Some examples follow.
    @example
    
    # 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
    
    
    # Write the MD5 hash of the encoded AVI file to stdout.
    
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    ffmpeg -i input.flv -f avi -y md5:
    @end example
    
    
    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.
    
    @section pipe
    
    UNIX pipe access protocol.
    
    Allow to read and write from UNIX pipes.
    
    The accepted syntax is:
    @example
    pipe:[@var{number}]
    @end example
    
    @var{number} is the number corresponding to the file descriptor of the
    
    pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).  If @var{number}
    is not specified, by default the stdout file descriptor will be used
    for writing, stdin for reading.
    
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    For example to read from stdin with @file{ffmpeg}:
    @example
    cat test.wav | ffmpeg -i pipe:0
    
    # ...this is the same as...
    
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    cat test.wav | ffmpeg -i pipe:
    @end example
    
    For writing to stdout with @file{ffmpeg}:
    @example
    ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
    
    # ...this is the same as...
    
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    ffmpeg -i test.wav -f avi pipe: | cat > test.avi
    @end example
    
    
    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.
    
    @section rtmp
    
    Real-Time Messaging Protocol.
    
    The Real-Time Messaging Protocol (RTMP) is used for streaming multime‐
    dia content across a TCP/IP network.
    
    The required syntax is:
    @example
    rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
    @end example
    
    
    The accepted parameters are:
    
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    @table @option
    
    @item server
    
    The address of the RTMP server.
    
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    @item port
    
    The number of the TCP port to use (by default is 1935).
    
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    @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
    
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    (e.g. @file{/ondemand/}, @file{/flash/live/}, etc.).
    
    @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:".
    
    @end table
    
    For example to read with @file{ffplay} a multimedia resource named
    "sample" from the application "vod" from an RTMP server "myserver":
    @example
    ffplay rtmp://myserver/vod/sample
    @end example
    
    @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
    
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    configuration. You need to explicitely configure the build with
    "--enable-librtmp". If enabled this will replace the native RTMP
    protocol.
    
    This protocol provides most client functions and a few server
    functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
    encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
    variants of these encrypted types (RTMPTE, RTMPTS).
    
    The required syntax is:
    @example
    @var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
    @end example
    
    where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
    "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
    @var{server}, @var{port}, @var{app} and @var{playpath} have the same
    
    meaning as specified for the RTMP native protocol.
    
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    @var{options} contains a list of space-separated options of the form
    @var{key}=@var{val}.
    
    
    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
    @file{ffmpeg}:
    @example
    ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
    @end example
    
    To play the same stream using @file{ffplay}:
    @example
    ffplay "rtmp://myserver/live/mystream live=1"
    @end example
    
    @section rtp
    
    Real-Time Protocol.
    
    
    @section rtsp
    
    RTSP is not technically a protocol handler in libavformat, it is a demuxer
    and muxer. The demuxer supports both normal RTSP (with data transferred
    over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
    data transferred over RDT).
    
    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}).
    
    The required syntax for a RTSP url is:
    @example
    rtsp://@var{hostname}[:@var{port}]/@var{path}[?@var{options}]
    @end example
    
    @var{options} is a @code{&}-separated list. The following options
    are supported:
    
    @table @option
    
    @item udp
    Use UDP as lower transport protocol.
    
    @item tcp
    Use TCP (interleaving within the RTSP control channel) as lower
    transport protocol.
    
    @item multicast
    Use UDP multicast as lower transport protocol.
    
    @item http
    Use HTTP tunneling as lower transport protocol, which is useful for
    passing proxies.
    @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.
    
    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.
    
    When watching multi-bitrate Real-RTSP streams with @file{ffplay}, 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}.
    
    Example command lines:
    
    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
    @end example
    
    To watch a stream tunneled over HTTP:
    
    @example
    ffplay rtsp://server/video.mp4?http
    @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
    @end example
    
    
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    @section sap
    
    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
    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:
    @example
    sap://@var{destination}[:@var{port}][?@var{options}]
    @end example
    
    The RTP packets are sent to @var{destination} on port @var{port},
    or to port 5004 if no port is specified.
    @var{options} is a @code{&}-separated list. The following options
    are supported:
    
    @table @option
    
    @item announce_addr=@var{address}
    Specify the destination IP address for sending the announcements to.
    If omitted, the announcements are sent to the commonly used SAP
    announcement multicast address 224.2.127.254 (sap.mcast.net), or
    ff0e::2:7ffe if @var{destination} is an IPv6 address.
    
    @item announce_port=@var{port}
    Specify the port to send the announcements on, defaults to
    9875 if not specified.
    
    @item ttl=@var{ttl}
    Specify the time to live value for the announcements and RTP packets,
    defaults to 255.
    
    @item same_port=@var{0|1}
    If set to 1, send all RTP streams on the same port pair. If zero (the
    default), all streams are sent on unique ports, with each stream on a
    port 2 numbers higher than the previous.
    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
    on unique ports.
    
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    @end table
    
    Example command lines follow.
    
    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
    @end example
    
    
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    Similarly, for watching in ffplay:
    
    @example
    ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
    @end example
    
    And for watching in ffplay, over IPv6:
    
    @example
    ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
    @end example
    
    @subsection Demuxer
    
    The syntax for a SAP url given to the demuxer is:
    @example
    sap://[@var{address}][:@var{port}]
    @end example
    
    @var{address} is the multicast address to listen for announcements on,
    if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
    is the port that is listened on, 9875 if omitted.
    
    The demuxers listens for announcements on the given address and port.
    Once an announcement is received, it tries to receive that particular stream.
    
    Example command lines follow.
    
    To play back the first stream announced on the normal SAP multicast address:
    
    @example
    ffplay sap://
    @end example
    
    To play back the first stream announced on one the default IPv6 SAP multicast address:
    
    @example
    ffplay sap://[ff0e::2:7ffe]
    @end example
    
    
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    @section tcp
    
    Trasmission Control Protocol.
    
    @section udp
    
    User Datagram Protocol.
    
    
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    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 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 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.
    
    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.
    
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    @end table
    
    Some usage examples of the udp protocol with @file{ffmpeg} follow.
    
    To stream over UDP to a remote endpoint:
    @example
    ffmpeg -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
    @end example
    
    To receive over UDP from a remote endpoint:
    @example
    ffmpeg -i udp://[@var{multicast-address}]:@var{port}
    @end example
    
    
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    @c man end PROTOCOLS