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diff --git a/ffmpeg/doc/protocols.texi b/ffmpeg/doc/protocols.texi deleted file mode 100644 index 57f9266..0000000 --- a/ffmpeg/doc/protocols.texi +++ /dev/null @@ -1,1104 +0,0 @@ -@chapter Protocols -@c man begin PROTOCOLS - -Protocols are configured elements in FFmpeg that enable access to -resources that require specific protocols. - -When you configure your FFmpeg build, all the supported protocols are -enabled by default. You can list all available ones using the -configure option "--list-protocols". - -You can disable all the protocols using the configure option -"--disable-protocols", and selectively enable a protocol using the -option "--enable-protocol=@var{PROTOCOL}", or you can disable a -particular protocol using the option -"--disable-protocol=@var{PROTOCOL}". - -The option "-protocols" of the ff* tools will display the list of -supported protocols. - -A description of the currently available protocols follows. - -@section bluray - -Read BluRay playlist. - -The accepted options are: -@table @option - -@item angle -BluRay angle - -@item chapter -Start chapter (1...N) - -@item playlist -Playlist to read (BDMV/PLAYLIST/?????.mpls) - -@end table - -Examples: - -Read longest playlist from BluRay mounted to /mnt/bluray: -@example -bluray:/mnt/bluray -@end example - -Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: -@example --playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray -@end example - -@section cache - -Caching wrapper for input stream. - -Cache the input stream to temporary file. It brings seeking capability to live streams. - -@example -cache:@var{URL} -@end example - -@section concat - -Physical concatenation protocol. - -Allow to read and seek from many resource in sequence as if they were -a unique resource. - -A URL accepted by this protocol has the syntax: -@example -concat:@var{URL1}|@var{URL2}|...|@var{URLN} -@end example - -where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the -resource to be concatenated, each one possibly specifying a distinct -protocol. - -For example to read a sequence of files @file{split1.mpeg}, -@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the -command: -@example -ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg -@end example - -Note that you may need to escape the character "|" which is special for -many shells. - -@section crypto - -AES-encrypted stream reading protocol. - -The accepted options are: -@table @option -@item key -Set the AES decryption key binary block from given hexadecimal representation. - -@item iv -Set the AES decryption initialization vector binary block from given hexadecimal representation. -@end table - -Accepted URL formats: -@example -crypto:@var{URL} -crypto+@var{URL} -@end example - -@section data - -Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}. - -For example, to convert a GIF file given inline with @command{ffmpeg}: -@example -ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png -@end example - -@section file - -File access protocol. - -Allow to read from or read to a file. - -For example to read from a file @file{input.mpeg} with @command{ffmpeg} -use the command: -@example -ffmpeg -i file:input.mpeg output.mpeg -@end example - -The ff* tools default to the file protocol, that is a resource -specified with the name "FILE.mpeg" is interpreted as the URL -"file:FILE.mpeg". - -This protocol accepts the following options: - -@table @option -@item truncate -Truncate existing files on write, if set to 1. A value of 0 prevents -truncating. Default value is 1. - -@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 for files on slow medium. -@end table - -@section ftp - -FTP (File Transfer Protocol). - -Allow to read from or write to remote resources using FTP protocol. - -Following syntax is required. -@example -ftp://[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 ftp-anonymous-password -Password used when login as anonymous user. Typically an e-mail address -should be used. - -@item ftp-write-seekable -Control seekability of connection during encoding. If set to 1 the -resource is supposed to be seekable, if set to 0 it is assumed not -to be seekable. Default value is 0. -@end table - -NOTE: Protocol can be used as output, but it is recommended to not do -it, unless special care is taken (tests, customized server configuration -etc.). Different FTP servers behave in different way during seek -operation. ff* tools may produce incomplete content due to server limitations. - -@section gopher - -Gopher protocol. - -@section hls - -Read Apple HTTP Live Streaming compliant segmented stream as -a uniform one. The M3U8 playlists describing the segments can be -remote HTTP resources or local files, accessed using the standard -file protocol. -The nested protocol is declared by specifying -"+@var{proto}" after the hls URI scheme name, where @var{proto} -is either "file" or "http". - -@example -hls+http://host/path/to/remote/resource.m3u8 -hls+file://path/to/local/resource.m3u8 -@end example - -Using this protocol is discouraged - the hls demuxer should work -just as well (if not, please report the issues) and is more complete. -To use the hls demuxer instead, simply use the direct URLs to the -m3u8 files. - -@section http - -HTTP (Hyper Text Transfer Protocol). - -This protocol accepts the following options. - -@table @option -@item seekable -Control seekability of connection. If set to 1 the resource is -supposed to be seekable, if set to 0 it is assumed not to be seekable, -if set to -1 it will try to autodetect if it is seekable. Default -value is -1. - -@item chunked_post -If set to 1 use chunked transfer-encoding for posts, default is 1. - -@item headers -Set custom HTTP headers, can override built in default headers. The -value must be a string encoding the headers. - -@item content_type -Force a content type. - -@item user-agent -Override User-Agent header. If not specified the protocol will use a -string describing the libavformat build. - -@item multiple_requests -Use persistent connections if set to 1. By default it is 0. - -@item post_data -Set custom HTTP post data. - -@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 mime_type -Set MIME type. - -@item icy -If set to 1 request ICY (SHOUTcast) metadata from the server. If the server -supports this, the metadata has to be retrieved by the application by reading -the @option{icy_metadata_headers} and @option{icy_metadata_packet} options. -The default is 0. - -@item icy_metadata_headers -If the server supports ICY metadata, this contains the ICY specific HTTP reply -headers, separated with newline characters. - -@item icy_metadata_packet -If the server supports ICY metadata, and @option{icy} was set to 1, this -contains the last non-empty metadata packet sent by the server. - -@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 -delimited by a newline character. -@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 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. - -Allow to read and write from UNIX pipes. - -The accepted syntax is: -@example -pipe:[@var{number}] -@end example - -@var{number} is the number corresponding to the file descriptor of the -pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number} -is not specified, by default the stdout file descriptor will be used -for writing, stdin for reading. - -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 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 libssh - -Secure File Transfer Protocol via libssh - -Allow to 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. - -@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 the local RTP port value plus 1. -@end enumerate - -@section rtsp - -RTSP is not technically a protocol handler in libavformat, it is a demuxer -and muxer. The demuxer supports both normal RTSP (with data transferred -over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with -data transferred over RDT). - -The muxer can be used to send a stream using RTSP ANNOUNCE to a server -supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's -@uref{http://github.com/revmischa/rtsp-server, RTSP server}). - -The required syntax for a RTSP url is: -@example -rtsp://@var{hostname}[:@var{port}]/@var{path} -@end example - -The following options (set on the @command{ffmpeg}/@command{ffplay} command -line, or set in code via @code{AVOption}s or in @code{avformat_open_input}), -are supported: - -Flags for @code{rtsp_transport}: - -@table @option - -@item udp -Use 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 @code{tcp} and @code{udp} options are supported. - -Flags for @code{rtsp_flags}: - -@table @option -@item filter_src -Accept packets only from negotiated peer address and port. -@item listen -Act as a server, listening for an incoming connection. -@end table - -When receiving data over UDP, the demuxer tries to reorder received packets -(since they may arrive out of order, or packets may get lost totally). 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}. - -Example command lines: - -To 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 - -To watch a stream tunneled over HTTP: - -@example -ffplay -rtsp_transport http rtsp://server/video.mp4 -@end example - -To send a stream in realtime to a RTSP server, for others to watch: - -@example -ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp -@end example - -To receive a stream in realtime: - -@example -ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output} -@end example - -@table @option -@item stimeout -Socket IO timeout in micro seconds. -@end table - -@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 tcp - -Trasmission 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{microseconds} -Set listen timeout, expressed in microseconds. -@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 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 socket buffer size in bytes. This is used both for the -receiving and the sending buffer size. - -@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. -@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 |