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1448 lines
40 KiB
Plaintext
1448 lines
40 KiB
Plaintext
@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 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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@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-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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This protocol accepts the following options:
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@table @option
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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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@end table
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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 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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@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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@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
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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
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updates.
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@item cookies
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Set the cookies to be sent in future requests. The format of each cookie is the
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same as the value of a Set-Cookie HTTP response field. Multiple cookies can be
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delimited by a newline character.
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@item offset
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Set initial byte offset.
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@item end_offset
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Try to limit the request to bytes preceding this offset.
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@item method
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When used as a client option it sets the HTTP method for the request.
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When used as a server option it sets the HTTP method that is going to be
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expected from the client(s).
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If the expected and the received HTTP method do not match the client will
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be given a Bad Request response.
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When unset the HTTP method is not checked for now. This will be replaced by
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autodetection in the future.
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@item listen
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If set to 1 enables experimental HTTP server. This can be used to send data when
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used as an output option, or read data from a client with HTTP POST when used as
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an input option.
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If set to 2 enables experimental multi-client HTTP server. This is not yet implemented
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in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
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@example
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# 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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# Client side (receiving):
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ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
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# Client can also be done with wget:
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wget http://@var{server}:@var{port} -O somefile.ogg
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# Server side (receiving):
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ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
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# Client side (sending):
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ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
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# Client can also be done with wget:
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wget --post-file=somefile.ogg http://@var{server}:@var{port}
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@end example
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@end table
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@subsection HTTP Cookies
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Some HTTP requests will be denied unless cookie values are passed in with the
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request. The @option{cookies} option allows these cookies to be specified. At
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the very least, each cookie must specify a value along with a path and domain.
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HTTP requests that match both the domain and path will automatically include the
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cookie value in the HTTP Cookie header field. Multiple cookies can be delimited
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by a newline.
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The required syntax to play a stream specifying a cookie is:
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@example
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ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
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@end example
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@section Icecast
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Icecast protocol (stream to Icecast servers)
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This protocol accepts the following options:
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@table @option
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@item ice_genre
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Set the stream genre.
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@item ice_name
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Set the stream name.
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@item ice_description
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Set the stream description.
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@item ice_url
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Set the stream website URL.
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@item ice_public
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Set if the stream should be public.
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The default is 0 (not public).
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@item user_agent
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Override the User-Agent header. If not specified a string of the form
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"Lavf/<version>" will be used.
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@item password
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Set the Icecast mountpoint password.
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@item content_type
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Set the stream content type. This must be set if it is different from
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audio/mpeg.
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@item legacy_icecast
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This enables support for Icecast versions < 2.4.0, that do not support the
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HTTP PUT method but the SOURCE method.
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@end table
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@example
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icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
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@end example
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@section mmst
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MMS (Microsoft Media Server) protocol over TCP.
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@section mmsh
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MMS (Microsoft Media Server) protocol over HTTP.
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The required syntax is:
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@example
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mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
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@end example
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@section md5
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MD5 output protocol.
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Computes the MD5 hash of the data to be written, and on close writes
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this to the designated output or stdout if none is specified. It can
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be used to test muxers without writing an actual file.
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Some examples follow.
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@example
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# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
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ffmpeg -i input.flv -f avi -y md5:output.avi.md5
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# Write the MD5 hash of the encoded AVI file to stdout.
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ffmpeg -i input.flv -f avi -y md5:
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@end example
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Note that some formats (typically MOV) require the output protocol to
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be seekable, so they will fail with the MD5 output protocol.
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@section pipe
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UNIX pipe access protocol.
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Read and write from UNIX pipes.
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The accepted syntax is:
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@example
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pipe:[@var{number}]
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@end example
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@var{number} is the number corresponding to the file descriptor of the
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pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
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is not specified, by default the stdout file descriptor will be used
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for writing, stdin for reading.
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For example to read from stdin with @command{ffmpeg}:
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@example
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cat test.wav | ffmpeg -i pipe:0
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# ...this is the same as...
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cat test.wav | ffmpeg -i pipe:
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@end example
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For writing to stdout with @command{ffmpeg}:
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@example
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ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
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# ...this is the same as...
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ffmpeg -i test.wav -f avi pipe: | cat > test.avi
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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 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 if data transmission is slow.
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@end table
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Note that some formats (typically MOV), require the output protocol to
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be seekable, so they will fail with the pipe output protocol.
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@section prompeg
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Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
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The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism
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for MPEG-2 Transport Streams sent over RTP.
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This protocol must be used in conjunction with the @code{rtp_mpegts} muxer and
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the @code{rtp} protocol.
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The required syntax is:
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@example
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-f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port}
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@end example
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The destination UDP ports are @code{port + 2} for the column FEC stream
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and @code{port + 4} for the row FEC stream.
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This protocol accepts the following options:
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@table @option
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@item l=@var{n}
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The number of columns (4-20, LxD <= 100)
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@item d=@var{n}
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The number of rows (4-20, LxD <= 100)
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@end table
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Example usage:
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@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 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.
|
|
@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
|
|
|
|
@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.
|
|
@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 GnuTLS,
|
|
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 64KB. 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}
|
|
Choose the local IP address. This is useful e.g. if sending multicast
|
|
and the host has multiple interfaces, where the user can choose
|
|
which interface to send on by specifying the IP address of that interface.
|
|
|
|
@item pkt_size=@var{size}
|
|
Set the size in bytes of UDP packets.
|
|
|
|
@item reuse=@var{1|0}
|
|
Explicitly allow or disallow reusing UDP sockets.
|
|
|
|
@item ttl=@var{ttl}
|
|
Set the time to live value (for multicast only).
|
|
|
|
@item connect=@var{1|0}
|
|
Initialize the UDP socket with @code{connect()}. In this case, the
|
|
destination address can't be changed with ff_udp_set_remote_url later.
|
|
If the destination address isn't known at the start, this option can
|
|
be specified in ff_udp_set_remote_url, too.
|
|
This allows finding out the source address for the packets with getsockname,
|
|
and makes writes return with AVERROR(ECONNREFUSED) if "destination
|
|
unreachable" is received.
|
|
For receiving, this gives the benefit of only receiving packets from
|
|
the specified peer address/port.
|
|
|
|
@item sources=@var{address}[,@var{address}]
|
|
Only receive packets sent to the multicast group from one of the
|
|
specified sender IP addresses.
|
|
|
|
@item block=@var{address}[,@var{address}]
|
|
Ignore packets sent to the multicast group from the specified
|
|
sender IP addresses.
|
|
|
|
@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
|
|
|
|
@c man end PROTOCOLS
|