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1889 lines
57 KiB
Plaintext
Executable File
1889 lines
57 KiB
Plaintext
Executable File
@chapter Protocol Options
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@c man begin PROTOCOL OPTIONS
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The libavformat library provides some generic global options, which
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can be set on all the protocols. In addition each protocol may support
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so-called private options, which are specific for that component.
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Options may be set by specifying -@var{option} @var{value} in the
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FFmpeg tools, or by setting the value explicitly in the
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@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
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for programmatic use.
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The list of supported options follows:
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@table @option
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@item protocol_whitelist @var{list} (@emph{input})
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Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
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prefixed by "-" are disabled.
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All protocols are allowed by default but protocols used by an another
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protocol (nested protocols) are restricted to a per protocol subset.
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@end table
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@c man end PROTOCOL OPTIONS
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@chapter Protocols
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@c man begin PROTOCOLS
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Protocols are configured elements in FFmpeg that enable access to
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resources that require specific protocols.
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When you configure your FFmpeg 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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All protocols accept the following options:
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@table @option
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@item rw_timeout
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Maximum time to wait for (network) read/write operations to complete,
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in microseconds.
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@end table
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A description of the currently available protocols follows.
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@section amqp
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Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based
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publish-subscribe communication protocol.
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FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A separate
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AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ.
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After starting the broker, an FFmpeg client may stream data to the broker using
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the command:
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@example
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ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@@]hostname[:port]
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@end example
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Where hostname and port (default is 5672) is the address of the broker. The
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client may also set a user/password for authentication. The default for both
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fields is "guest".
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Muliple subscribers may stream from the broker using the command:
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@example
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ffplay amqp://[[user]:[password]@@]hostname[:port]
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@end example
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In RabbitMQ all data published to the broker flows through a specific exchange,
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and each subscribing client has an assigned queue/buffer. When a packet arrives
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at an exchange, it may be copied to a client's queue depending on the exchange
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and routing_key fields.
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The following options are supported:
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@table @option
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@item exchange
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Sets the exchange to use on the broker. RabbitMQ has several predefined
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exchanges: "amq.direct" is the default exchange, where the publisher and
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subscriber must have a matching routing_key; "amq.fanout" is the same as a
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broadcast operation (i.e. the data is forwarded to all queues on the fanout
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exchange independent of the routing_key); and "amq.topic" is similar to
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"amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ
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documentation).
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@item routing_key
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Sets the routing key. The default value is "amqp". The routing key is used on
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the "amq.direct" and "amq.topic" exchanges to decide whether packets are written
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to the queue of a subscriber.
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@item pkt_size
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Maximum size of each packet sent/received to the broker. Default is 131072.
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Minimum is 4096 and max is any large value (representable by an int). When
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receiving packets, this sets an internal buffer size in FFmpeg. It should be
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equal to or greater than the size of the published packets to the broker. Otherwise
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the received message may be truncated causing decoding errors.
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@item connection_timeout
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The timeout in seconds during the initial connection to the broker. The
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default value is rw_timeout, or 5 seconds if rw_timeout is not set.
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@item delivery_mode @var{mode}
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Sets the delivery mode of each message sent to broker.
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The following values are accepted:
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@table @samp
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@item persistent
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Delivery mode set to "persistent" (2). This is the default value.
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Messages may be written to the broker's disk depending on its setup.
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@item non-persistent
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Delivery mode set to "non-persistent" (1).
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Messages will stay in broker's memory unless the broker is under memory
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pressure.
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@end table
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@end table
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@section async
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Asynchronous data filling wrapper for input stream.
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Fill data in a background thread, to decouple I/O operation from demux thread.
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@example
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async:@var{URL}
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async:http://host/resource
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async:cache:http://host/resource
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@end example
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@section bluray
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Read BluRay playlist.
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The accepted options are:
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@table @option
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@item angle
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BluRay angle
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@item chapter
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Start chapter (1...N)
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@item playlist
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Playlist to read (BDMV/PLAYLIST/?????.mpls)
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@end table
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Examples:
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Read longest playlist from BluRay mounted to /mnt/bluray:
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@example
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bluray:/mnt/bluray
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@end example
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Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
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@example
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-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
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@end example
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@section cache
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Caching wrapper for input stream.
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Cache the input stream to temporary file. It brings seeking capability to live streams.
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@example
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cache:@var{URL}
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@end example
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@section concat
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Physical concatenation protocol.
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Read and seek from many resources 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{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 crypto
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AES-encrypted stream reading protocol.
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The accepted options are:
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@table @option
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@item key
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Set the AES decryption key binary block from given hexadecimal representation.
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@item iv
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Set the AES decryption initialization vector binary block from given hexadecimal representation.
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@end table
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Accepted URL formats:
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@example
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crypto:@var{URL}
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crypto+@var{URL}
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@end example
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@section data
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Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
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For example, to convert a GIF file given inline with @command{ffmpeg}:
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@example
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ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
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@end example
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@section file
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File access protocol.
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Read from or write to a file.
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A file URL can have the form:
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@example
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file:@var{filename}
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@end example
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where @var{filename} is the path of the file to read.
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An URL that does not have a protocol prefix will be assumed to be a
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file URL. Depending on the build, an URL that looks like a Windows
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path with the drive letter at the beginning will also be assumed to be
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a file URL (usually not the case in builds for unix-like systems).
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For example to read from a file @file{input.mpeg} with @command{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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This protocol accepts the following options:
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@table @option
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@item truncate
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Truncate existing files on write, if set to 1. A value of 0 prevents
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truncating. Default value is 1.
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@item blocksize
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Set I/O operation maximum block size, in bytes. Default value is
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@code{INT_MAX}, which results in not limiting the requested block size.
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Setting this value reasonably low improves user termination request reaction
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time, which is valuable for files on slow medium.
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@item follow
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If set to 1, the protocol will retry reading at the end of the file, allowing
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reading files that still are being written. In order for this to terminate,
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you either need to use the rw_timeout option, or use the interrupt callback
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(for API users).
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@item seekable
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Controls if seekability is advertised on the file. 0 means non-seekable, -1
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means auto (seekable for normal files, non-seekable for named pipes).
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Many demuxers handle seekable and non-seekable resources differently,
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overriding this might speed up opening certain files at the cost of losing some
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features (e.g. accurate seeking).
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@end table
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@section ftp
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FTP (File Transfer Protocol).
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Read from or write to remote resources using FTP protocol.
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Following syntax is required.
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@example
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ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
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@end example
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This protocol accepts the following options.
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@table @option
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@item timeout
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Set timeout in microseconds of socket I/O operations used by the underlying low level
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operation. By default it is set to -1, which means that the timeout is
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not specified.
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@item ftp-user
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Set a user to be used for authenticating to the FTP server. This is overridden by the
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user in the FTP URL.
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@item ftp-password
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Set a password to be used for authenticating to the FTP server. This is overridden by
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the password in the FTP URL, or by @option{ftp-anonymous-password} if no user is set.
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@item ftp-anonymous-password
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Password used when login as anonymous user. Typically an e-mail address
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should be used.
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@item ftp-write-seekable
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Control seekability of connection during encoding. If set to 1 the
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resource is supposed to be seekable, if set to 0 it is assumed not
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to be seekable. Default value is 0.
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@end table
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NOTE: Protocol can be used as output, but it is recommended to not do
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it, unless special care is taken (tests, customized server configuration
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etc.). Different FTP servers behave in different way during seek
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operation. ff* tools may produce incomplete content due to server limitations.
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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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This protocol accepts the following options:
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@table @option
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@item seekable
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Control seekability of connection. If set to 1 the resource is
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supposed to be seekable, if set to 0 it is assumed not to be seekable,
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if set to -1 it will try to autodetect if it is seekable. Default
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value is -1.
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@item chunked_post
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If set to 1 use chunked Transfer-Encoding for posts, default is 1.
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@item content_type
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Set a specific content type for the POST messages or for listen mode.
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@item http_proxy
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set HTTP proxy to tunnel through e.g. http://example.com:1234
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@item headers
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Set custom HTTP headers, can override built in default headers. The
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value must be a string encoding the headers.
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@item multiple_requests
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Use persistent connections if set to 1, default is 0.
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@item post_data
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Set custom HTTP post data.
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@item referer
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Set the Referer header. Include 'Referer: URL' header in HTTP request.
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@item user_agent
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Override the User-Agent header. If not specified the protocol will use a
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string describing the libavformat build. ("Lavf/<version>")
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@item user-agent
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This is a deprecated option, you can use user_agent instead it.
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@item timeout
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Set timeout in microseconds of socket I/O operations used by the underlying low level
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operation. By default it is set to -1, which means that the timeout is
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not specified.
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||
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@item reconnect_at_eof
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If set then eof is treated like an error and causes reconnection, this is useful
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for live / endless streams.
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@item reconnect_streamed
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If set then even streamed/non seekable streams will be reconnected on errors.
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@item reconnect_delay_max
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Sets the maximum delay in seconds after which to give up reconnecting
|
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@item mime_type
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Export the MIME type.
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||
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@item http_version
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Exports the HTTP response version number. Usually "1.0" or "1.1".
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@item icy
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If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
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supports this, the metadata has to be retrieved by the application by reading
|
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the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
|
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The default is 1.
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||
|
||
@item icy_metadata_headers
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If the server supports ICY metadata, this contains the ICY-specific HTTP reply
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headers, separated by newline characters.
|
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|
||
@item icy_metadata_packet
|
||
If the server supports ICY metadata, and @option{icy} was set to 1, this
|
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contains the last non-empty metadata packet sent by the server. It should be
|
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polled in regular intervals by applications interested in mid-stream metadata
|
||
updates.
|
||
|
||
@item cookies
|
||
Set the cookies to be sent in future requests. The format of each cookie is the
|
||
same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
|
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delimited by a newline character.
|
||
|
||
@item offset
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||
Set initial byte offset.
|
||
|
||
@item end_offset
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||
Try to limit the request to bytes preceding this offset.
|
||
|
||
@item method
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When used as a client option it sets the HTTP method for the request.
|
||
|
||
When used as a server option it sets the HTTP method that is going to be
|
||
expected from the client(s).
|
||
If the expected and the received HTTP method do not match the client will
|
||
be given a Bad Request response.
|
||
When unset the HTTP method is not checked for now. This will be replaced by
|
||
autodetection in the future.
|
||
|
||
@item listen
|
||
If set to 1 enables experimental HTTP server. This can be used to send data when
|
||
used as an output option, or read data from a client with HTTP POST when used as
|
||
an input option.
|
||
If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
|
||
in ffmpeg.c and thus must not be used as a command line option.
|
||
@example
|
||
# Server side (sending):
|
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ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
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|
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# Client side (receiving):
|
||
ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
|
||
|
||
# Client can also be done with wget:
|
||
wget http://@var{server}:@var{port} -O somefile.ogg
|
||
|
||
# Server side (receiving):
|
||
ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
|
||
|
||
# Client side (sending):
|
||
ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
|
||
|
||
# Client can also be done with wget:
|
||
wget --post-file=somefile.ogg http://@var{server}:@var{port}
|
||
@end example
|
||
|
||
@item send_expect_100
|
||
Send an Expect: 100-continue header for POST. If set to 1 it will send, if set
|
||
to 0 it won't, if set to -1 it will try to send if it is applicable. Default
|
||
value is -1.
|
||
|
||
@end table
|
||
|
||
@subsection HTTP Cookies
|
||
|
||
Some HTTP requests will be denied unless cookie values are passed in with the
|
||
request. The @option{cookies} option allows these cookies to be specified. At
|
||
the very least, each cookie must specify a value along with a path and domain.
|
||
HTTP requests that match both the domain and path will automatically include the
|
||
cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
|
||
by a newline.
|
||
|
||
The required syntax to play a stream specifying a cookie is:
|
||
@example
|
||
ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
|
||
@end example
|
||
|
||
@section Icecast
|
||
|
||
Icecast protocol (stream to Icecast servers)
|
||
|
||
This protocol accepts the following options:
|
||
|
||
@table @option
|
||
@item ice_genre
|
||
Set the stream genre.
|
||
|
||
@item ice_name
|
||
Set the stream name.
|
||
|
||
@item ice_description
|
||
Set the stream description.
|
||
|
||
@item ice_url
|
||
Set the stream website URL.
|
||
|
||
@item ice_public
|
||
Set if the stream should be public.
|
||
The default is 0 (not public).
|
||
|
||
@item user_agent
|
||
Override the User-Agent header. If not specified a string of the form
|
||
"Lavf/<version>" will be used.
|
||
|
||
@item password
|
||
Set the Icecast mountpoint password.
|
||
|
||
@item content_type
|
||
Set the stream content type. This must be set if it is different from
|
||
audio/mpeg.
|
||
|
||
@item legacy_icecast
|
||
This enables support for Icecast versions < 2.4.0, that do not support the
|
||
HTTP PUT method but the SOURCE method.
|
||
|
||
@item tls
|
||
Establish a TLS (HTTPS) connection to Icecast.
|
||
|
||
@end table
|
||
|
||
@example
|
||
icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
|
||
@end example
|
||
|
||
@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
|
||
|
||
@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.
|
||
|
||
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
|
||
|
||
# Write the MD5 hash of the encoded AVI file to stdout.
|
||
ffmpeg -i input.flv -f avi -y md5:
|
||
@end example
|
||
|
||
Note that some formats (typically MOV) require the output protocol to
|
||
be seekable, so they will fail with the MD5 output protocol.
|
||
|
||
@section pipe
|
||
|
||
UNIX pipe access protocol.
|
||
|
||
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.
|
||
|
||
For example to read from stdin with @command{ffmpeg}:
|
||
@example
|
||
cat test.wav | ffmpeg -i pipe:0
|
||
# ...this is the same as...
|
||
cat test.wav | ffmpeg -i pipe:
|
||
@end example
|
||
|
||
For writing to stdout with @command{ffmpeg}:
|
||
@example
|
||
ffmpeg -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
|
||
@end example
|
||
|
||
This protocol accepts the following options:
|
||
|
||
@table @option
|
||
@item blocksize
|
||
Set I/O operation maximum block size, in bytes. Default value is
|
||
@code{INT_MAX}, which results in not limiting the requested block size.
|
||
Setting this value reasonably low improves user termination request reaction
|
||
time, which is valuable if data transmission is slow.
|
||
@end table
|
||
|
||
Note that some formats (typically MOV), require the output protocol to
|
||
be seekable, so they will fail with the pipe output protocol.
|
||
|
||
@section prompeg
|
||
|
||
Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
|
||
|
||
The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
|
||
for MPEG-2 Transport Streams sent over RTP.
|
||
|
||
This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
|
||
the @code{rtp} protocol.
|
||
|
||
The required syntax is:
|
||
@example
|
||
-f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
|
||
@end example
|
||
|
||
The destination UDP ports are @code{port + 2} for the column FEC stream
|
||
and @code{port + 4} for the row FEC stream.
|
||
|
||
This protocol accepts the following options:
|
||
@table @option
|
||
|
||
@item l=@var{n}
|
||
The number of columns (4-20, LxD <= 100)
|
||
|
||
@item d=@var{n}
|
||
The number of rows (4-20, LxD <= 100)
|
||
|
||
@end table
|
||
|
||
Example usage:
|
||
|
||
@example
|
||
-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port}
|
||
@end example
|
||
|
||
@section rtmp
|
||
|
||
Real-Time Messaging Protocol.
|
||
|
||
The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
|
||
content across a TCP/IP network.
|
||
|
||
The required syntax is:
|
||
@example
|
||
rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
|
||
@end example
|
||
|
||
The accepted parameters are:
|
||
@table @option
|
||
|
||
@item username
|
||
An optional username (mostly for publishing).
|
||
|
||
@item password
|
||
An optional password (mostly for publishing).
|
||
|
||
@item server
|
||
The address of the RTMP server.
|
||
|
||
@item port
|
||
The number of the TCP port to use (by default is 1935).
|
||
|
||
@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.). 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:". 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. (When publishing, the default is FMLE/3.0 (compatible;
|
||
<libavformat version>).)
|
||
|
||
@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{ffplay} a multimedia resource named
|
||
"sample" from the application "vod" from an RTMP server "myserver":
|
||
@example
|
||
ffplay rtmp://myserver/vod/sample
|
||
@end example
|
||
|
||
To publish to a password protected server, passing the playpath and
|
||
app names separately:
|
||
@example
|
||
ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
|
||
@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 libsmbclient
|
||
|
||
libsmbclient permits one to manipulate CIFS/SMB network resources.
|
||
|
||
Following syntax is required.
|
||
|
||
@example
|
||
smb://[[domain:]user[:password@@]]server[/share[/path[/file]]]
|
||
@end example
|
||
|
||
This protocol accepts the following options.
|
||
|
||
@table @option
|
||
@item timeout
|
||
Set timeout in milliseconds of socket I/O operations used by the underlying
|
||
low level operation. By default it is set to -1, which means that the timeout
|
||
is not specified.
|
||
|
||
@item truncate
|
||
Truncate existing files on write, if set to 1. A value of 0 prevents
|
||
truncating. Default value is 1.
|
||
|
||
@item workgroup
|
||
Set the workgroup used for making connections. By default workgroup is not specified.
|
||
|
||
@end table
|
||
|
||
For more information see: @url{http://www.samba.org/}.
|
||
|
||
@section libssh
|
||
|
||
Secure File Transfer Protocol via libssh
|
||
|
||
Read from or write to remote resources using SFTP protocol.
|
||
|
||
Following syntax is required.
|
||
|
||
@example
|
||
sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg
|
||
@end example
|
||
|
||
This protocol accepts the following options.
|
||
|
||
@table @option
|
||
@item timeout
|
||
Set timeout of socket I/O operations used by the underlying low level
|
||
operation. By default it is set to -1, which means that the timeout
|
||
is not specified.
|
||
|
||
@item truncate
|
||
Truncate existing files on write, if set to 1. A value of 0 prevents
|
||
truncating. Default value is 1.
|
||
|
||
@item private_key
|
||
Specify the path of the file containing private key to use during authorization.
|
||
By default libssh searches for keys in the @file{~/.ssh/} directory.
|
||
|
||
@end table
|
||
|
||
Example: Play a file stored on remote server.
|
||
|
||
@example
|
||
ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg
|
||
@end example
|
||
|
||
@section librtmp 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 explicitly 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.
|
||
@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.
|
||
|
||
For example, to stream a file in real-time to an RTMP server using
|
||
@command{ffmpeg}:
|
||
@example
|
||
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
|
||
@end example
|
||
|
||
To play the same stream using @command{ffplay}:
|
||
@example
|
||
ffplay "rtmp://myserver/live/mystream live=1"
|
||
@end example
|
||
|
||
@section rtp
|
||
|
||
Real-time Transport Protocol.
|
||
|
||
The required syntax for an RTP URL is:
|
||
rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
|
||
|
||
@var{port} specifies the RTP port to use.
|
||
|
||
The following URL options are supported:
|
||
|
||
@table @option
|
||
|
||
@item ttl=@var{n}
|
||
Set the TTL (Time-To-Live) value (for multicast only).
|
||
|
||
@item rtcpport=@var{n}
|
||
Set the remote RTCP port to @var{n}.
|
||
|
||
@item localrtpport=@var{n}
|
||
Set the local RTP port to @var{n}.
|
||
|
||
@item localrtcpport=@var{n}'
|
||
Set the local RTCP port to @var{n}.
|
||
|
||
@item pkt_size=@var{n}
|
||
Set max packet size (in bytes) to @var{n}.
|
||
|
||
@item connect=0|1
|
||
Do a @code{connect()} on the UDP socket (if set to 1) or not (if set
|
||
to 0).
|
||
|
||
@item sources=@var{ip}[,@var{ip}]
|
||
List allowed source IP addresses.
|
||
|
||
@item block=@var{ip}[,@var{ip}]
|
||
List disallowed (blocked) source IP addresses.
|
||
|
||
@item write_to_source=0|1
|
||
Send packets to the source address of the latest received packet (if
|
||
set to 1) or to a default remote address (if set to 0).
|
||
|
||
@item localport=@var{n}
|
||
Set the local RTP port to @var{n}.
|
||
|
||
This is a deprecated option. Instead, @option{localrtpport} should be
|
||
used.
|
||
|
||
@end table
|
||
|
||
Important notes:
|
||
|
||
@enumerate
|
||
|
||
@item
|
||
If @option{rtcpport} is not set the RTCP port will be set to the RTP
|
||
port value plus 1.
|
||
|
||
@item
|
||
If @option{localrtpport} (the local RTP port) is not set any available
|
||
port will be used for the local RTP and RTCP ports.
|
||
|
||
@item
|
||
If @option{localrtcpport} (the local RTCP port) is not set it will be
|
||
set to the local RTP port value plus 1.
|
||
@end enumerate
|
||
|
||
@section rtsp
|
||
|
||
Real-Time Streaming Protocol.
|
||
|
||
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
|
||
@uref{https://github.com/revmischa/rtsp-server, RTSP server}).
|
||
|
||
The required syntax for a RTSP url is:
|
||
@example
|
||
rtsp://@var{hostname}[:@var{port}]/@var{path}
|
||
@end example
|
||
|
||
Options can be set on the @command{ffmpeg}/@command{ffplay} command
|
||
line, or set in code via @code{AVOption}s or in
|
||
@code{avformat_open_input}.
|
||
|
||
The following options are supported.
|
||
|
||
@table @option
|
||
@item initial_pause
|
||
Do not start playing the stream immediately if set to 1. Default value
|
||
is 0.
|
||
|
||
@item rtsp_transport
|
||
Set RTSP transport protocols.
|
||
|
||
It accepts the following values:
|
||
@table @samp
|
||
@item udp
|
||
Use UDP as lower transport protocol.
|
||
|
||
@item tcp
|
||
Use TCP (interleaving within the RTSP control channel) as lower
|
||
transport protocol.
|
||
|
||
@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.
|
||
@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 @samp{tcp} and @samp{udp} options are supported.
|
||
|
||
@item rtsp_flags
|
||
Set RTSP flags.
|
||
|
||
The following values are accepted:
|
||
@table @samp
|
||
@item filter_src
|
||
Accept packets only from negotiated peer address and port.
|
||
@item listen
|
||
Act as a server, listening for an incoming connection.
|
||
@item prefer_tcp
|
||
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
|
||
@end table
|
||
|
||
Default value is @samp{none}.
|
||
|
||
@item allowed_media_types
|
||
Set media types to accept from the server.
|
||
|
||
The following flags are accepted:
|
||
@table @samp
|
||
@item video
|
||
@item audio
|
||
@item data
|
||
@end table
|
||
|
||
By default it accepts all media types.
|
||
|
||
@item min_port
|
||
Set minimum local UDP port. Default value is 5000.
|
||
|
||
@item max_port
|
||
Set maximum local UDP port. Default value is 65000.
|
||
|
||
@item timeout
|
||
Set maximum timeout (in seconds) to wait for incoming connections.
|
||
|
||
A value of -1 means infinite (default). This option implies the
|
||
@option{rtsp_flags} set to @samp{listen}.
|
||
|
||
@item reorder_queue_size
|
||
Set number of packets to buffer for handling of reordered packets.
|
||
|
||
@item stimeout
|
||
Set socket TCP I/O timeout in microseconds.
|
||
|
||
@item user-agent
|
||
Override User-Agent header. If not specified, it defaults to the
|
||
libavformat identifier string.
|
||
@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). 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 @command{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}.
|
||
|
||
@subsection Examples
|
||
|
||
The following examples all make use of the @command{ffplay} and
|
||
@command{ffmpeg} tools.
|
||
|
||
@itemize
|
||
@item
|
||
Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
|
||
@example
|
||
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
|
||
@end example
|
||
|
||
@item
|
||
Watch a stream tunneled over HTTP:
|
||
@example
|
||
ffplay -rtsp_transport http rtsp://server/video.mp4
|
||
@end example
|
||
|
||
@item
|
||
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
|
||
|
||
@item
|
||
Receive a stream in realtime:
|
||
@example
|
||
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
|
||
@end example
|
||
@end itemize
|
||
|
||
@section sap
|
||
|
||
Session Announcement Protocol (RFC 2974). This is not technically a
|
||
protocol handler in libavformat, it is a muxer and demuxer.
|
||
It is used for signalling of RTP streams, by announcing the SDP for the
|
||
streams regularly on a separate port.
|
||
|
||
@subsection Muxer
|
||
|
||
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.
|
||
The RTP stack in libavformat for receiving requires all streams to be sent
|
||
on unique ports.
|
||
@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
|
||
|
||
Similarly, for watching in @command{ffplay}:
|
||
|
||
@example
|
||
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
|
||
@end example
|
||
|
||
And for watching in @command{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
|
||
|
||
@section sctp
|
||
|
||
Stream Control Transmission Protocol.
|
||
|
||
The accepted URL syntax is:
|
||
@example
|
||
sctp://@var{host}:@var{port}[?@var{options}]
|
||
@end example
|
||
|
||
The protocol accepts the following options:
|
||
@table @option
|
||
@item listen
|
||
If set to any value, listen for an incoming connection. Outgoing connection is done by default.
|
||
|
||
@item max_streams
|
||
Set the maximum number of streams. By default no limit is set.
|
||
@end table
|
||
|
||
@section srt
|
||
|
||
Haivision Secure Reliable Transport Protocol via libsrt.
|
||
|
||
The supported syntax for a SRT URL is:
|
||
@example
|
||
srt://@var{hostname}:@var{port}[?@var{options}]
|
||
@end example
|
||
|
||
@var{options} contains a list of &-separated options of the form
|
||
@var{key}=@var{val}.
|
||
|
||
or
|
||
|
||
@example
|
||
@var{options} srt://@var{hostname}:@var{port}
|
||
@end example
|
||
|
||
@var{options} contains a list of '-@var{key} @var{val}'
|
||
options.
|
||
|
||
This protocol accepts the following options.
|
||
|
||
@table @option
|
||
@item connect_timeout=@var{milliseconds}
|
||
Connection timeout; SRT cannot connect for RTT > 1500 msec
|
||
(2 handshake exchanges) with the default connect timeout of
|
||
3 seconds. This option applies to the caller and rendezvous
|
||
connection modes. The connect timeout is 10 times the value
|
||
set for the rendezvous mode (which can be used as a
|
||
workaround for this connection problem with earlier versions).
|
||
|
||
@item ffs=@var{bytes}
|
||
Flight Flag Size (Window Size), in bytes. FFS is actually an
|
||
internal parameter and you should set it to not less than
|
||
@option{recv_buffer_size} and @option{mss}. The default value
|
||
is relatively large, therefore unless you set a very large receiver buffer,
|
||
you do not need to change this option. Default value is 25600.
|
||
|
||
@item inputbw=@var{bytes/seconds}
|
||
Sender nominal input rate, in bytes per seconds. Used along with
|
||
@option{oheadbw}, when @option{maxbw} is set to relative (0), to
|
||
calculate maximum sending rate when recovery packets are sent
|
||
along with the main media stream:
|
||
@option{inputbw} * (100 + @option{oheadbw}) / 100
|
||
if @option{inputbw} is not set while @option{maxbw} is set to
|
||
relative (0), the actual input rate is evaluated inside
|
||
the library. Default value is 0.
|
||
|
||
@item iptos=@var{tos}
|
||
IP Type of Service. Applies to sender only. Default value is 0xB8.
|
||
|
||
@item ipttl=@var{ttl}
|
||
IP Time To Live. Applies to sender only. Default value is 64.
|
||
|
||
@item latency=@var{microseconds}
|
||
Timestamp-based Packet Delivery Delay.
|
||
Used to absorb bursts of missed packet retransmissions.
|
||
This flag sets both @option{rcvlatency} and @option{peerlatency}
|
||
to the same value. Note that prior to version 1.3.0
|
||
this is the only flag to set the latency, however
|
||
this is effectively equivalent to setting @option{peerlatency},
|
||
when side is sender and @option{rcvlatency}
|
||
when side is receiver, and the bidirectional stream
|
||
sending is not supported.
|
||
|
||
@item listen_timeout=@var{microseconds}
|
||
Set socket listen timeout.
|
||
|
||
@item maxbw=@var{bytes/seconds}
|
||
Maximum sending bandwidth, in bytes per seconds.
|
||
-1 infinite (CSRTCC limit is 30mbps)
|
||
0 relative to input rate (see @option{inputbw})
|
||
>0 absolute limit value
|
||
Default value is 0 (relative)
|
||
|
||
@item mode=@var{caller|listener|rendezvous}
|
||
Connection mode.
|
||
@option{caller} opens client connection.
|
||
@option{listener} starts server to listen for incoming connections.
|
||
@option{rendezvous} use Rendez-Vous connection mode.
|
||
Default value is caller.
|
||
|
||
@item mss=@var{bytes}
|
||
Maximum Segment Size, in bytes. Used for buffer allocation
|
||
and rate calculation using a packet counter assuming fully
|
||
filled packets. The smallest MSS between the peers is
|
||
used. This is 1500 by default in the overall internet.
|
||
This is the maximum size of the UDP packet and can be
|
||
only decreased, unless you have some unusual dedicated
|
||
network settings. Default value is 1500.
|
||
|
||
@item nakreport=@var{1|0}
|
||
If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
|
||
periodically until a lost packet is retransmitted or
|
||
intentionally dropped. Default value is 1.
|
||
|
||
@item oheadbw=@var{percents}
|
||
Recovery bandwidth overhead above input rate, in percents.
|
||
See @option{inputbw}. Default value is 25%.
|
||
|
||
@item passphrase=@var{string}
|
||
HaiCrypt Encryption/Decryption Passphrase string, length
|
||
from 10 to 79 characters. The passphrase is the shared
|
||
secret between the sender and the receiver. It is used
|
||
to generate the Key Encrypting Key using PBKDF2
|
||
(Password-Based Key Derivation Function). It is used
|
||
only if @option{pbkeylen} is non-zero. It is used on
|
||
the receiver only if the received data is encrypted.
|
||
The configured passphrase cannot be recovered (write-only).
|
||
|
||
@item enforced_encryption=@var{1|0}
|
||
If true, both connection parties must have the same password
|
||
set (including empty, that is, with no encryption). If the
|
||
password doesn't match or only one side is unencrypted,
|
||
the connection is rejected. Default is true.
|
||
|
||
@item kmrefreshrate=@var{packets}
|
||
The number of packets to be transmitted after which the
|
||
encryption key is switched to a new key. Default is -1.
|
||
-1 means auto (0x1000000 in srt library). The range for
|
||
this option is integers in the 0 - @code{INT_MAX}.
|
||
|
||
@item kmpreannounce=@var{packets}
|
||
The interval between when a new encryption key is sent and
|
||
when switchover occurs. This value also applies to the
|
||
subsequent interval between when switchover occurs and
|
||
when the old encryption key is decommissioned. Default is -1.
|
||
-1 means auto (0x1000 in srt library). The range for
|
||
this option is integers in the 0 - @code{INT_MAX}.
|
||
|
||
@item payload_size=@var{bytes}
|
||
Sets the maximum declared size of a packet transferred
|
||
during the single call to the sending function in Live
|
||
mode. Use 0 if this value isn't used (which is default in
|
||
file mode).
|
||
Default is -1 (automatic), which typically means MPEG-TS;
|
||
if you are going to use SRT
|
||
to send any different kind of payload, such as, for example,
|
||
wrapping a live stream in very small frames, then you can
|
||
use a bigger maximum frame size, though not greater than
|
||
1456 bytes.
|
||
|
||
@item pkt_size=@var{bytes}
|
||
Alias for @samp{payload_size}.
|
||
|
||
@item peerlatency=@var{microseconds}
|
||
The latency value (as described in @option{rcvlatency}) that is
|
||
set by the sender side as a minimum value for the receiver.
|
||
|
||
@item pbkeylen=@var{bytes}
|
||
Sender encryption key length, in bytes.
|
||
Only can be set to 0, 16, 24 and 32.
|
||
Enable sender encryption if not 0.
|
||
Not required on receiver (set to 0),
|
||
key size obtained from sender in HaiCrypt handshake.
|
||
Default value is 0.
|
||
|
||
@item rcvlatency=@var{microseconds}
|
||
The time that should elapse since the moment when the
|
||
packet was sent and the moment when it's delivered to
|
||
the receiver application in the receiving function.
|
||
This time should be a buffer time large enough to cover
|
||
the time spent for sending, unexpectedly extended RTT
|
||
time, and the time needed to retransmit the lost UDP
|
||
packet. The effective latency value will be the maximum
|
||
of this options' value and the value of @option{peerlatency}
|
||
set by the peer side. Before version 1.3.0 this option
|
||
is only available as @option{latency}.
|
||
|
||
@item recv_buffer_size=@var{bytes}
|
||
Set UDP receive buffer size, expressed in bytes.
|
||
|
||
@item send_buffer_size=@var{bytes}
|
||
Set UDP send buffer size, expressed in bytes.
|
||
|
||
@item timeout=@var{microseconds}
|
||
Set raise error timeouts for read, write and connect operations. Note that the
|
||
SRT library has internal timeouts which can be controlled separately, the
|
||
value set here is only a cap on those.
|
||
|
||
@item tlpktdrop=@var{1|0}
|
||
Too-late Packet Drop. When enabled on receiver, it skips
|
||
missing packets that have not been delivered in time and
|
||
delivers the following packets to the application when
|
||
their time-to-play has come. It also sends a fake ACK to
|
||
the sender. When enabled on sender and enabled on the
|
||
receiving peer, the sender drops the older packets that
|
||
have no chance of being delivered in time. It was
|
||
automatically enabled in the sender if the receiver
|
||
supports it.
|
||
|
||
@item sndbuf=@var{bytes}
|
||
Set send buffer size, expressed in bytes.
|
||
|
||
@item rcvbuf=@var{bytes}
|
||
Set receive buffer size, expressed in bytes.
|
||
|
||
Receive buffer must not be greater than @option{ffs}.
|
||
|
||
@item lossmaxttl=@var{packets}
|
||
The value up to which the Reorder Tolerance may grow. When
|
||
Reorder Tolerance is > 0, then packet loss report is delayed
|
||
until that number of packets come in. Reorder Tolerance
|
||
increases every time a "belated" packet has come, but it
|
||
wasn't due to retransmission (that is, when UDP packets tend
|
||
to come out of order), with the difference between the latest
|
||
sequence and this packet's sequence, and not more than the
|
||
value of this option. By default it's 0, which means that this
|
||
mechanism is turned off, and the loss report is always sent
|
||
immediately upon experiencing a "gap" in sequences.
|
||
|
||
@item minversion
|
||
The minimum SRT version that is required from the peer. A connection
|
||
to a peer that does not satisfy the minimum version requirement
|
||
will be rejected.
|
||
|
||
The version format in hex is 0xXXYYZZ for x.y.z in human readable
|
||
form.
|
||
|
||
@item streamid=@var{string}
|
||
A string limited to 512 characters that can be set on the socket prior
|
||
to connecting. This stream ID will be able to be retrieved by the
|
||
listener side from the socket that is returned from srt_accept and
|
||
was connected by a socket with that set stream ID. SRT does not enforce
|
||
any special interpretation of the contents of this string.
|
||
This option doesn’t make sense in Rendezvous connection; the result
|
||
might be that simply one side will override the value from the other
|
||
side and it’s the matter of luck which one would win
|
||
|
||
@item smoother=@var{live|file}
|
||
The type of Smoother used for the transmission for that socket, which
|
||
is responsible for the transmission and congestion control. The Smoother
|
||
type must be exactly the same on both connecting parties, otherwise
|
||
the connection is rejected.
|
||
|
||
@item messageapi=@var{1|0}
|
||
When set, this socket uses the Message API, otherwise it uses Buffer
|
||
API. Note that in live mode (see @option{transtype}) there’s only
|
||
message API available. In File mode you can chose to use one of two modes:
|
||
|
||
Stream API (default, when this option is false). In this mode you may
|
||
send as many data as you wish with one sending instruction, or even use
|
||
dedicated functions that read directly from a file. The internal facility
|
||
will take care of any speed and congestion control. When receiving, you
|
||
can also receive as many data as desired, the data not extracted will be
|
||
waiting for the next call. There is no boundary between data portions in
|
||
the Stream mode.
|
||
|
||
Message API. In this mode your single sending instruction passes exactly
|
||
one piece of data that has boundaries (a message). Contrary to Live mode,
|
||
this message may span across multiple UDP packets and the only size
|
||
limitation is that it shall fit as a whole in the sending buffer. The
|
||
receiver shall use as large buffer as necessary to receive the message,
|
||
otherwise the message will not be given up. When the message is not
|
||
complete (not all packets received or there was a packet loss) it will
|
||
not be given up.
|
||
|
||
@item transtype=@var{live|file}
|
||
Sets the transmission type for the socket, in particular, setting this
|
||
option sets multiple other parameters to their default values as required
|
||
for a particular transmission type.
|
||
|
||
live: Set options as for live transmission. In this mode, you should
|
||
send by one sending instruction only so many data that fit in one UDP packet,
|
||
and limited to the value defined first in @option{payload_size} (1316 is
|
||
default in this mode). There is no speed control in this mode, only the
|
||
bandwidth control, if configured, in order to not exceed the bandwidth with
|
||
the overhead transmission (retransmitted and control packets).
|
||
|
||
file: Set options as for non-live transmission. See @option{messageapi}
|
||
for further explanations
|
||
|
||
@item linger=@var{seconds}
|
||
The number of seconds that the socket waits for unsent data when closing.
|
||
Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
|
||
seconds in file mode). The range for this option is integers in the
|
||
0 - @code{INT_MAX}.
|
||
|
||
@end table
|
||
|
||
For more information see: @url{https://github.com/Haivision/srt}.
|
||
|
||
@section srtp
|
||
|
||
Secure Real-time Transport Protocol.
|
||
|
||
The accepted options are:
|
||
@table @option
|
||
@item srtp_in_suite
|
||
@item srtp_out_suite
|
||
Select input and output encoding suites.
|
||
|
||
Supported values:
|
||
@table @samp
|
||
@item AES_CM_128_HMAC_SHA1_80
|
||
@item SRTP_AES128_CM_HMAC_SHA1_80
|
||
@item AES_CM_128_HMAC_SHA1_32
|
||
@item SRTP_AES128_CM_HMAC_SHA1_32
|
||
@end table
|
||
|
||
@item srtp_in_params
|
||
@item srtp_out_params
|
||
Set input and output encoding parameters, which are expressed by a
|
||
base64-encoded representation of a binary block. The first 16 bytes of
|
||
this binary block are used as master key, the following 14 bytes are
|
||
used as master salt.
|
||
@end table
|
||
|
||
@section subfile
|
||
|
||
Virtually extract a segment of a file or another stream.
|
||
The underlying stream must be seekable.
|
||
|
||
Accepted options:
|
||
@table @option
|
||
@item start
|
||
Start offset of the extracted segment, in bytes.
|
||
@item end
|
||
End offset of the extracted segment, in bytes.
|
||
If set to 0, extract till end of file.
|
||
@end table
|
||
|
||
Examples:
|
||
|
||
Extract a chapter from a DVD VOB file (start and end sectors obtained
|
||
externally and multiplied by 2048):
|
||
@example
|
||
subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
|
||
@end example
|
||
|
||
Play an AVI file directly from a TAR archive:
|
||
@example
|
||
subfile,,start,183241728,end,366490624,,:archive.tar
|
||
@end example
|
||
|
||
Play a MPEG-TS file from start offset till end:
|
||
@example
|
||
subfile,,start,32815239,end,0,,:video.ts
|
||
@end example
|
||
|
||
@section tee
|
||
|
||
Writes the output to multiple protocols. The individual outputs are separated
|
||
by |
|
||
|
||
@example
|
||
tee:file://path/to/local/this.avi|file://path/to/local/that.avi
|
||
@end example
|
||
|
||
@section tcp
|
||
|
||
Transmission Control Protocol.
|
||
|
||
The required syntax for a TCP url is:
|
||
@example
|
||
tcp://@var{hostname}:@var{port}[?@var{options}]
|
||
@end example
|
||
|
||
@var{options} contains a list of &-separated options of the form
|
||
@var{key}=@var{val}.
|
||
|
||
The list of supported options follows.
|
||
|
||
@table @option
|
||
@item listen=@var{1|0}
|
||
Listen for an incoming connection. Default value is 0.
|
||
|
||
@item timeout=@var{microseconds}
|
||
Set raise error timeout, expressed in microseconds.
|
||
|
||
This option is only relevant in read mode: if no data arrived in more
|
||
than this time interval, raise error.
|
||
|
||
@item listen_timeout=@var{milliseconds}
|
||
Set listen timeout, expressed in milliseconds.
|
||
|
||
@item recv_buffer_size=@var{bytes}
|
||
Set receive buffer size, expressed bytes.
|
||
|
||
@item send_buffer_size=@var{bytes}
|
||
Set send buffer size, expressed bytes.
|
||
|
||
@item tcp_nodelay=@var{1|0}
|
||
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
|
||
|
||
@item tcp_mss=@var{bytes}
|
||
Set maximum segment size for outgoing TCP packets, expressed in bytes.
|
||
@end table
|
||
|
||
The following example shows how to setup a listening TCP connection
|
||
with @command{ffmpeg}, which is then accessed with @command{ffplay}:
|
||
@example
|
||
ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
|
||
ffplay tcp://@var{hostname}:@var{port}
|
||
@end example
|
||
|
||
@section tls
|
||
|
||
Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
|
||
|
||
The required syntax for a TLS/SSL url is:
|
||
@example
|
||
tls://@var{hostname}:@var{port}[?@var{options}]
|
||
@end example
|
||
|
||
The following parameters can be set via command line options
|
||
(or in code via @code{AVOption}s):
|
||
|
||
@table @option
|
||
|
||
@item ca_file, cafile=@var{filename}
|
||
A file containing certificate authority (CA) root certificates to treat
|
||
as trusted. If the linked TLS library contains a default this might not
|
||
need to be specified for verification to work, but not all libraries and
|
||
setups have defaults built in.
|
||
The file must be in OpenSSL PEM format.
|
||
|
||
@item tls_verify=@var{1|0}
|
||
If enabled, try to verify the peer that we are communicating with.
|
||
Note, if using OpenSSL, this currently only makes sure that the
|
||
peer certificate is signed by one of the root certificates in the CA
|
||
database, but it does not validate that the certificate actually
|
||
matches the host name we are trying to connect to. (With other backends,
|
||
the host name is validated as well.)
|
||
|
||
This is disabled by default since it requires a CA database to be
|
||
provided by the caller in many cases.
|
||
|
||
@item cert_file, cert=@var{filename}
|
||
A file containing a certificate to use in the handshake with the peer.
|
||
(When operating as server, in listen mode, this is more often required
|
||
by the peer, while client certificates only are mandated in certain
|
||
setups.)
|
||
|
||
@item key_file, key=@var{filename}
|
||
A file containing the private key for the certificate.
|
||
|
||
@item listen=@var{1|0}
|
||
If enabled, listen for connections on the provided port, and assume
|
||
the server role in the handshake instead of the client role.
|
||
|
||
@end table
|
||
|
||
Example command lines:
|
||
|
||
To create a TLS/SSL server that serves an input stream.
|
||
|
||
@example
|
||
ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
|
||
@end example
|
||
|
||
To play back a stream from the TLS/SSL server using @command{ffplay}:
|
||
|
||
@example
|
||
ffplay tls://@var{hostname}:@var{port}
|
||
@end example
|
||
|
||
@section udp
|
||
|
||
User Datagram Protocol.
|
||
|
||
The required syntax for an UDP URL is:
|
||
@example
|
||
udp://@var{hostname}:@var{port}[?@var{options}]
|
||
@end example
|
||
|
||
@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
|
||
|
||
In case threading is enabled on the system, a circular buffer is used
|
||
to store the incoming data, which allows one to reduce loss of data due to
|
||
UDP socket buffer overruns. The @var{fifo_size} and
|
||
@var{overrun_nonfatal} options are related to this buffer.
|
||
|
||
The list of supported options follows.
|
||
|
||
@table @option
|
||
@item buffer_size=@var{size}
|
||
Set the UDP maximum socket buffer size in bytes. This is used to set either
|
||
the receive or send buffer size, depending on what the socket is used for.
|
||
Default is 32 KB for output, 384 KB for input. See also @var{fifo_size}.
|
||
|
||
@item bitrate=@var{bitrate}
|
||
If set to nonzero, the output will have the specified constant bitrate if the
|
||
input has enough packets to sustain it.
|
||
|
||
@item burst_bits=@var{bits}
|
||
When using @var{bitrate} this specifies the maximum number of bits in
|
||
packet bursts.
|
||
|
||
@item localport=@var{port}
|
||
Override the local UDP port to bind with.
|
||
|
||
@item localaddr=@var{addr}
|
||
Local IP address of a network interface used for sending packets or joining
|
||
multicast groups.
|
||
|
||
@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 from the specified addresses. In case of multicast,
|
||
also subscribe to multicast traffic coming from these addresses only.
|
||
|
||
@item block=@var{address}[,@var{address}]
|
||
Ignore packets sent from the specified addresses. In case of multicast, also
|
||
exclude the source addresses in the multicast subscription.
|
||
|
||
@item fifo_size=@var{units}
|
||
Set the UDP receiving circular buffer size, expressed as a number of
|
||
packets with size of 188 bytes. If not specified defaults to 7*4096.
|
||
|
||
@item overrun_nonfatal=@var{1|0}
|
||
Survive in case of UDP receiving circular buffer overrun. Default
|
||
value is 0.
|
||
|
||
@item timeout=@var{microseconds}
|
||
Set raise error timeout, expressed in microseconds.
|
||
|
||
This option is only relevant in read mode: if no data arrived in more
|
||
than this time interval, raise error.
|
||
|
||
@item broadcast=@var{1|0}
|
||
Explicitly allow or disallow UDP broadcasting.
|
||
|
||
Note that broadcasting may not work properly on networks having
|
||
a broadcast storm protection.
|
||
@end table
|
||
|
||
@subsection Examples
|
||
|
||
@itemize
|
||
@item
|
||
Use @command{ffmpeg} to stream over UDP to a remote endpoint:
|
||
@example
|
||
ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
|
||
@end example
|
||
|
||
@item
|
||
Use @command{ffmpeg} 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
|
||
|
||
@item
|
||
Use @command{ffmpeg} to receive over UDP from a remote endpoint:
|
||
@example
|
||
ffmpeg -i udp://[@var{multicast-address}]:@var{port} ...
|
||
@end example
|
||
@end itemize
|
||
|
||
@section unix
|
||
|
||
Unix local socket
|
||
|
||
The required syntax for a Unix socket URL is:
|
||
|
||
@example
|
||
unix://@var{filepath}
|
||
@end example
|
||
|
||
The following parameters can be set via command line options
|
||
(or in code via @code{AVOption}s):
|
||
|
||
@table @option
|
||
@item timeout
|
||
Timeout in ms.
|
||
@item listen
|
||
Create the Unix socket in listening mode.
|
||
@end table
|
||
|
||
@section zmq
|
||
|
||
ZeroMQ asynchronous messaging using the libzmq library.
|
||
|
||
This library supports unicast streaming to multiple clients without relying on
|
||
an external server.
|
||
|
||
The required syntax for streaming or connecting to a stream is:
|
||
@example
|
||
zmq:tcp://ip-address:port
|
||
@end example
|
||
|
||
Example:
|
||
Create a localhost stream on port 5555:
|
||
@example
|
||
ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
|
||
@end example
|
||
|
||
Multiple clients may connect to the stream using:
|
||
@example
|
||
ffplay zmq:tcp://127.0.0.1:5555
|
||
@end example
|
||
|
||
Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.
|
||
The server side binds to a port and publishes data. Clients connect to the
|
||
server (via IP address/port) and subscribe to the stream. The order in which
|
||
the server and client start generally does not matter.
|
||
|
||
ffmpeg must be compiled with the --enable-libzmq option to support
|
||
this protocol.
|
||
|
||
Options can be set on the @command{ffmpeg}/@command{ffplay} command
|
||
line. The following options are supported:
|
||
|
||
@table @option
|
||
|
||
@item pkt_size
|
||
Forces the maximum packet size for sending/receiving data. The default value is
|
||
131,072 bytes. On the server side, this sets the maximum size of sent packets
|
||
via ZeroMQ. On the clients, it sets an internal buffer size for receiving
|
||
packets. Note that pkt_size on the clients should be equal to or greater than
|
||
pkt_size on the server. Otherwise the received message may be truncated causing
|
||
decoding errors.
|
||
|
||
@end table
|
||
|
||
|
||
@c man end PROTOCOLS
|