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            "text": "# FFMPEG-PROTOCOLS (man)\n\n## NAME\n\nffmpeg-protocols - FFmpeg protocols\n\n## DESCRIPTION\n\nThis document describes the input and output protocols provided by the libavformat library.\n\n## Sections\n\n- **NAME**\n- **DESCRIPTION**\n- **PROTOCOL OPTIONS**\n- **PROTOCOLS**\n- **SEE ALSO**\n- **AUTHORS**\n\nUse structuredContent.sections for detailed options, examples, and full documentation.\n"
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        "summary": "ffmpeg-protocols - FFmpeg protocols",
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            {
                "name": "ffmpeg",
                "section": "1",
                "url": "https://www.chedong.com/phpMan.php/man/ffmpeg/1/json"
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                "name": "ffplay",
                "section": "1",
                "url": "https://www.chedong.com/phpMan.php/man/ffplay/1/json"
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            {
                "name": "ffprobe",
                "section": "1",
                "url": "https://www.chedong.com/phpMan.php/man/ffprobe/1/json"
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                "name": "libavformat",
                "section": "3",
                "url": "https://www.chedong.com/phpMan.php/man/libavformat/3/json"
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            {
                "name": "NAME",
                "lines": 2,
                "subsections": []
            },
            {
                "name": "DESCRIPTION",
                "lines": 2,
                "subsections": []
            },
            {
                "name": "PROTOCOL OPTIONS",
                "lines": 15,
                "subsections": []
            },
            {
                "name": "PROTOCOLS",
                "lines": 1565,
                "subsections": []
            },
            {
                "name": "SEE ALSO",
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                "subsections": []
            },
            {
                "name": "AUTHORS",
                "lines": 10,
                "subsections": []
            }
        ],
        "sections": {
            "NAME": {
                "content": "ffmpeg-protocols - FFmpeg protocols\n",
                "subsections": []
            },
            "DESCRIPTION": {
                "content": "This document describes the input and output protocols provided by the libavformat library.\n",
                "subsections": []
            },
            "PROTOCOL OPTIONS": {
                "content": "The libavformat library provides some generic global options, which can be set on all the\nprotocols. In addition each protocol may support so-called private options, which are\nspecific for that component.\n\nOptions may be set by specifying -option value in the FFmpeg tools, or by setting the value\nexplicitly in the \"AVFormatContext\" options or using the libavutil/opt.h API for programmatic\nuse.\n\nThe list of supported options follows:\n\nprotocolwhitelist list (input)\nSet a \",\"-separated list of allowed protocols. \"ALL\" matches all protocols. Protocols\nprefixed by \"-\" are disabled.  All protocols are allowed by default but protocols used by\nan another protocol (nested protocols) are restricted to a per protocol subset.\n",
                "subsections": []
            },
            "PROTOCOLS": {
                "content": "Protocols are configured elements in FFmpeg that enable access to resources that require\nspecific protocols.\n\nWhen you configure your FFmpeg build, all the supported protocols are enabled by default. You\ncan list all available ones using the configure option \"--list-protocols\".\n\nYou can disable all the protocols using the configure option \"--disable-protocols\", and\nselectively enable a protocol using the option \"--enable-protocol=PROTOCOL\", or you can\ndisable a particular protocol using the option \"--disable-protocol=PROTOCOL\".\n\nThe option \"-protocols\" of the ff* tools will display the list of supported protocols.\n\nAll protocols accept the following options:\n\nrwtimeout\nMaximum time to wait for (network) read/write operations to complete, in microseconds.\n\nA description of the currently available protocols follows.\n\namqp\nAdvanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based publish-subscribe\ncommunication protocol.\n\nFFmpeg must be compiled with --enable-librabbitmq to support AMQP. A separate AMQP broker\nmust also be run. An example open-source AMQP broker is RabbitMQ.\n\nAfter starting the broker, an FFmpeg client may stream data to the broker using the command:\n\nffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost]\n\nWhere hostname and port (default is 5672) is the address of the broker. The client may also\nset a user/password for authentication. The default for both fields is \"guest\". Name of\nvirtual host on broker can be set with vhost. The default value is \"/\".\n\nMuliple subscribers may stream from the broker using the command:\n\nffplay amqp://[[user]:[password]@]hostname[:port][/vhost]\n\nIn RabbitMQ all data published to the broker flows through a specific exchange, and each\nsubscribing client has an assigned queue/buffer. When a packet arrives at an exchange, it may\nbe copied to a client's queue depending on the exchange and routingkey fields.\n\nThe following options are supported:\n\nexchange\nSets the exchange to use on the broker. RabbitMQ has several predefined exchanges:\n\"amq.direct\" is the default exchange, where the publisher and subscriber must have a\nmatching routingkey; \"amq.fanout\" is the same as a broadcast operation (i.e. the data is\nforwarded to all queues on the fanout exchange independent of the routingkey); and\n\"amq.topic\" is similar to \"amq.direct\", but allows for more complex pattern matching\n(refer to the RabbitMQ documentation).\n\nroutingkey\nSets the routing key. The default value is \"amqp\". The routing key is used on the\n\"amq.direct\" and \"amq.topic\" exchanges to decide whether packets are written to the queue\nof a subscriber.\n\npktsize\nMaximum size of each packet sent/received to the broker. Default is 131072.  Minimum is\n4096 and max is any large value (representable by an int). When receiving packets, this\nsets an internal buffer size in FFmpeg. It should be equal to or greater than the size of\nthe published packets to the broker. Otherwise the received message may be truncated\ncausing decoding errors.\n\nconnectiontimeout\nThe timeout in seconds during the initial connection to the broker. The default value is\nrwtimeout, or 5 seconds if rwtimeout is not set.\n\ndeliverymode mode\nSets the delivery mode of each message sent to broker.  The following values are\naccepted:\n\npersistent\nDelivery mode set to \"persistent\" (2). This is the default value.  Messages may be\nwritten to the broker's disk depending on its setup.\n\nnon-persistent\nDelivery mode set to \"non-persistent\" (1).  Messages will stay in broker's memory\nunless the broker is under memory pressure.\n\nasync\nAsynchronous data filling wrapper for input stream.\n\nFill data in a background thread, to decouple I/O operation from demux thread.\n\nasync:<URL>\nasync:http://host/resource\nasync:cache:http://host/resource\n\nbluray\nRead BluRay playlist.\n\nThe accepted options are:\n\nangle\nBluRay angle\n\nchapter\nStart chapter (1...N)\n\nplaylist\nPlaylist to read (BDMV/PLAYLIST/?????.mpls)\n\nExamples:\n\nRead longest playlist from BluRay mounted to /mnt/bluray:\n\nbluray:/mnt/bluray\n\nRead angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:\n\n-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray\n\ncache\nCaching wrapper for input stream.\n\nCache the input stream to temporary file. It brings seeking capability to live streams.\n\nThe accepted options are:\n\nreadaheadlimit\nAmount in bytes that may be read ahead when seeking isn't supported. Range is -1 to\nINTMAX.  -1 for unlimited. Default is 65536.\n\nURL Syntax is\n\ncache:<URL>\n\nconcat\nPhysical concatenation protocol.\n\nRead and seek from many resources in sequence as if they were a unique resource.\n\nA URL accepted by this protocol has the syntax:\n\nconcat:<URL1>|<URL2>|...|<URLN>\n\nwhere URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one\npossibly specifying a distinct protocol.\n\nFor example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with ffplay use\nthe command:\n\nffplay concat:split1.mpeg\\|split2.mpeg\\|split3.mpeg\n\nNote that you may need to escape the character \"|\" which is special for many shells.\n\ncrypto\nAES-encrypted stream reading protocol.\n\nThe accepted options are:\n\nkey Set the AES decryption key binary block from given hexadecimal representation.\n\niv  Set the AES decryption initialization vector binary block from given hexadecimal\nrepresentation.\n\nAccepted URL formats:\n\ncrypto:<URL>\ncrypto+<URL>\n\ndata\nData in-line in the URI. See <http://en.wikipedia.org/wiki/DataURIscheme>.\n\nFor example, to convert a GIF file given inline with ffmpeg:\n\nffmpeg -i \"data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=\" smiley.png\n\nfile\nFile access protocol.\n\nRead from or write to a file.\n\nA file URL can have the form:\n\nfile:<filename>\n\nwhere filename is the path of the file to read.\n\nAn URL that does not have a protocol prefix will be assumed to be a file URL. Depending on\nthe build, an URL that looks like a Windows path with the drive letter at the beginning will\nalso be assumed to be a file URL (usually not the case in builds for unix-like systems).\n\nFor example to read from a file input.mpeg with ffmpeg use the command:\n\nffmpeg -i file:input.mpeg output.mpeg\n\nThis protocol accepts the following options:\n\ntruncate\nTruncate existing files on write, if set to 1. A value of 0 prevents truncating. Default\nvalue is 1.\n\nblocksize\nSet I/O operation maximum block size, in bytes. Default value is \"INTMAX\", which results\nin not limiting the requested block size.  Setting this value reasonably low improves\nuser termination request reaction time, which is valuable for files on slow medium.\n\nfollow\nIf set to 1, the protocol will retry reading at the end of the file, allowing reading\nfiles that still are being written. In order for this to terminate, you either need to\nuse the rwtimeout option, or use the interrupt callback (for API users).\n\nseekable\nControls if seekability is advertised on the file. 0 means non-seekable, -1 means auto\n(seekable for normal files, non-seekable for named pipes).\n\nMany demuxers handle seekable and non-seekable resources differently, overriding this\nmight speed up opening certain files at the cost of losing some features (e.g. accurate\nseeking).\n\nftp\nFTP (File Transfer Protocol).\n\nRead from or write to remote resources using FTP protocol.\n\nFollowing syntax is required.\n\nftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg\n\nThis protocol accepts the following options.\n\ntimeout\nSet timeout in microseconds of socket I/O operations used by the underlying low level\noperation. By default it is set to -1, which means that the timeout is not specified.\n\nftp-user\nSet a user to be used for authenticating to the FTP server. This is overridden by the\nuser in the FTP URL.\n\nftp-password\nSet a password to be used for authenticating to the FTP server. This is overridden by the\npassword in the FTP URL, or by ftp-anonymous-password if no user is set.\n\nftp-anonymous-password\nPassword used when login as anonymous user. Typically an e-mail address should be used.\n\nftp-write-seekable\nControl seekability of connection during encoding. If set to 1 the resource is supposed\nto be seekable, if set to 0 it is assumed not to be seekable. Default value is 0.\n\nNOTE: Protocol can be used as output, but it is recommended to not do it, unless special care\nis taken (tests, customized server configuration etc.). Different FTP servers behave in\ndifferent way during seek operation. ff* tools may produce incomplete content due to server\nlimitations.\n\ngopher\nGopher protocol.\n\ngophers\nGophers protocol.\n\nThe Gopher protocol with TLS encapsulation.\n\nhls\nRead Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8\nplaylists describing the segments can be remote HTTP resources or local files, accessed using\nthe standard file protocol.  The nested protocol is declared by specifying \"+proto\" after the\nhls URI scheme name, where proto is either \"file\" or \"http\".\n\nhls+http://host/path/to/remote/resource.m3u8\nhls+file://path/to/local/resource.m3u8\n\nUsing this protocol is discouraged - the hls demuxer should work just as well (if not, please\nreport the issues) and is more complete.  To use the hls demuxer instead, simply use the\ndirect URLs to the m3u8 files.\n\nhttp\nHTTP (Hyper Text Transfer Protocol).\n\nThis protocol accepts the following options:\n\nseekable\nControl seekability of connection. If set to 1 the resource is supposed to be seekable,\nif set to 0 it is assumed not to be seekable, if set to -1 it will try to autodetect if\nit is seekable. Default value is -1.\n\nchunkedpost\nIf set to 1 use chunked Transfer-Encoding for posts, default is 1.\n\ncontenttype\nSet a specific content type for the POST messages or for listen mode.\n\nhttpproxy\nset HTTP proxy to tunnel through e.g. http://example.com:1234\n\nheaders\nSet custom HTTP headers, can override built in default headers. The value must be a\nstring encoding the headers.\n\nmultiplerequests\nUse persistent connections if set to 1, default is 0.\n\npostdata\nSet custom HTTP post data.\n\nreferer\nSet the Referer header. Include 'Referer: URL' header in HTTP request.\n\nuseragent\nOverride the User-Agent header. If not specified the protocol will use a string\ndescribing the libavformat build. (\"Lavf/<version>\")\n\nuser-agent\nThis is a deprecated option, you can use useragent instead it.\n\nreconnectateof\nIf set then eof is treated like an error and causes reconnection, this is useful for live\n/ endless streams.\n\nreconnectstreamed\nIf set then even streamed/non seekable streams will be reconnected on errors.\n\nreconnectonnetworkerror\nReconnect automatically in case of TCP/TLS errors during connect.\n\nreconnectonhttperror\nA comma separated list of HTTP status codes to reconnect on. The list can include\nspecific status codes (e.g. '503') or the strings '4xx' / '5xx'.\n\nreconnectdelaymax\nSets the maximum delay in seconds after which to give up reconnecting\n\nmimetype\nExport the MIME type.\n\nhttpversion\nExports the HTTP response version number. Usually \"1.0\" or \"1.1\".\n\nicy If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports\nthis, the metadata has to be retrieved by the application by reading the\nicymetadataheaders and icymetadatapacket options.  The default is 1.\n\nicymetadataheaders\nIf the server supports ICY metadata, this contains the ICY-specific HTTP reply headers,\nseparated by newline characters.\n\nicymetadatapacket\nIf the server supports ICY metadata, and icy was set to 1, this contains the last non-\nempty metadata packet sent by the server. It should be polled in regular intervals by\napplications interested in mid-stream metadata updates.\n\ncookies\nSet the cookies to be sent in future requests. The format of each cookie is the same as\nthe value of a Set-Cookie HTTP response field. Multiple cookies can be delimited by a\nnewline character.\n\noffset\nSet initial byte offset.\n\nendoffset\nTry to limit the request to bytes preceding this offset.\n\nmethod\nWhen used as a client option it sets the HTTP method for the request.\n\nWhen used as a server option it sets the HTTP method that is going to be expected from\nthe client(s).  If the expected and the received HTTP method do not match the client will\nbe given a Bad Request response.  When unset the HTTP method is not checked for now. This\nwill be replaced by autodetection in the future.\n\nlisten\nIf set to 1 enables experimental HTTP server. This can be used to send data when used as\nan output option, or read data from a client with HTTP POST when used as an input option.\nIf set to 2 enables experimental multi-client HTTP server. This is not yet implemented in\nffmpeg.c and thus must not be used as a command line option.\n\n# Server side (sending):\nffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>\n\n# Client side (receiving):\nffmpeg -i http://<server>:<port> -c copy somefile.ogg\n\n# Client can also be done with wget:\nwget http://<server>:<port> -O somefile.ogg\n\n# Server side (receiving):\nffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg\n\n# Client side (sending):\nffmpeg -i somefile.ogg -chunkedpost 0 -c copy -f ogg http://<server>:<port>\n\n# Client can also be done with wget:\nwget --post-file=somefile.ogg http://<server>:<port>\n\nsendexpect100\nSend an Expect: 100-continue header for POST. If set to 1 it will send, if set to 0 it\nwon't, if set to -1 it will try to send if it is applicable. Default value is -1.\n\nauthtype\nSet HTTP authentication type. No option for Digest, since this method requires getting\nnonce parameters from the server first and can't be used straight away like Basic.\n\nnone\nChoose the HTTP authentication type automatically. This is the default.\n\nbasic\nChoose the HTTP basic authentication.\n\nBasic authentication sends a Base64-encoded string that contains a user name and\npassword for the client. Base64 is not a form of encryption and should be considered\nthe same as sending the user name and password in clear text (Base64 is a reversible\nencoding).  If a resource needs to be protected, strongly consider using an\nauthentication scheme other than basic authentication. HTTPS/TLS should be used with\nbasic authentication.  Without these additional security enhancements, basic\nauthentication should not be used to protect sensitive or valuable information.\n\nHTTP Cookies\n\nSome HTTP requests will be denied unless cookie values are passed in with the request. The\ncookies option allows these cookies to be specified. At the very least, each cookie must\nspecify a value along with a path and domain.  HTTP requests that match both the domain and\npath will automatically include the cookie value in the HTTP Cookie header field. Multiple\ncookies can be delimited by a newline.\n\nThe required syntax to play a stream specifying a cookie is:\n\nffplay -cookies \"nlqptid=nltid=tsn; path=/; domain=somedomain.com;\" http://somedomain.com/somestream.m3u8\n\nIcecast\nIcecast protocol (stream to Icecast servers)\n\nThis protocol accepts the following options:\n\nicegenre\nSet the stream genre.\n\nicename\nSet the stream name.\n\nicedescription\nSet the stream description.\n\niceurl\nSet the stream website URL.\n\nicepublic\nSet if the stream should be public.  The default is 0 (not public).\n\nuseragent\nOverride the User-Agent header. If not specified a string of the form \"Lavf/<version>\"\nwill be used.\n\npassword\nSet the Icecast mountpoint password.\n\ncontenttype\nSet the stream content type. This must be set if it is different from audio/mpeg.\n\nlegacyicecast\nThis enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT\nmethod but the SOURCE method.\n\ntls Establish a TLS (HTTPS) connection to Icecast.\n\nicecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>\n\nmmst\nMMS (Microsoft Media Server) protocol over TCP.\n\nmmsh\nMMS (Microsoft Media Server) protocol over HTTP.\n\nThe required syntax is:\n\nmmsh://<server>[:<port>][/<app>][/<playpath>]\n\nmd5\nMD5 output protocol.\n\nComputes the MD5 hash of the data to be written, and on close writes this to the designated\noutput or stdout if none is specified. It can be used to test muxers without writing an\nactual file.\n\nSome examples follow.\n\n# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.\nffmpeg -i input.flv -f avi -y md5:output.avi.md5\n\n# Write the MD5 hash of the encoded AVI file to stdout.\nffmpeg -i input.flv -f avi -y md5:\n\nNote that some formats (typically MOV) require the output protocol to be seekable, so they\nwill fail with the MD5 output protocol.\n\npipe\nUNIX pipe access protocol.\n\nRead and write from UNIX pipes.\n\nThe accepted syntax is:\n\npipe:[<number>]\n\nnumber is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1\nfor stdout, 2 for stderr).  If number is not specified, by default the stdout file descriptor\nwill be used for writing, stdin for reading.\n\nFor example to read from stdin with ffmpeg:\n\ncat test.wav | ffmpeg -i pipe:0\n# ...this is the same as...\ncat test.wav | ffmpeg -i pipe:\n\nFor writing to stdout with ffmpeg:\n\nffmpeg -i test.wav -f avi pipe:1 | cat > test.avi\n# ...this is the same as...\nffmpeg -i test.wav -f avi pipe: | cat > test.avi\n\nThis protocol accepts the following options:\n\nblocksize\nSet I/O operation maximum block size, in bytes. Default value is \"INTMAX\", which results\nin not limiting the requested block size.  Setting this value reasonably low improves\nuser termination request reaction time, which is valuable if data transmission is slow.\n\nNote that some formats (typically MOV), require the output protocol to be seekable, so they\nwill fail with the pipe output protocol.\n\nprompeg\nPro-MPEG Code of Practice #3 Release 2 FEC protocol.\n\nThe Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism for MPEG-2\nTransport Streams sent over RTP.\n\nThis protocol must be used in conjunction with the \"rtpmpegts\" muxer and the \"rtp\" protocol.\n\nThe required syntax is:\n\n-f rtpmpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>\n\nThe destination UDP ports are \"port + 2\" for the column FEC stream and \"port + 4\" for the row\nFEC stream.\n\nThis protocol accepts the following options:\n\nl=n The number of columns (4-20, LxD <= 100)\n\nd=n The number of rows (4-20, LxD <= 100)\n\nExample usage:\n\n-f rtpmpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>\n\nrist\nReliable Internet Streaming Transport protocol\n\nThe accepted options are:\n\nristprofile\nSupported values:\n\nsimple\nmain\nThis one is default.\n\nadvanced\nbuffersize\nSet internal RIST buffer size in milliseconds for retransmission of data.  Default value\nis 0 which means the librist default (1 sec). Maximum value is 30 seconds.\n\npktsize\nSet maximum packet size for sending data. 1316 by default.\n\nloglevel\nSet loglevel for RIST logging messages. You only need to set this if you explicitly want\nto enable debug level messages or packet loss simulation, otherwise the regular loglevel\nis respected.\n\nsecret\nSet override of encryption secret, by default is unset.\n\nencryption\nSet encryption type, by default is disabled.  Acceptable values are 128 and 256.\n\nrtmp\nReal-Time Messaging Protocol.\n\nThe Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a\nTCP/IP network.\n\nThe required syntax is:\n\nrtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]\n\nThe accepted parameters are:\n\nusername\nAn optional username (mostly for publishing).\n\npassword\nAn optional password (mostly for publishing).\n\nserver\nThe address of the RTMP server.\n\nport\nThe number of the TCP port to use (by default is 1935).\n\napp It is the name of the application to access. It usually corresponds to the path where the\napplication is installed on the RTMP server (e.g. /ondemand/, /flash/live/, etc.). You\ncan override the value parsed from the URI through the \"rtmpapp\" option, too.\n\nplaypath\nIt is the path or name of the resource to play with reference to the application\nspecified in app, may be prefixed by \"mp4:\". You can override the value parsed from the\nURI through the \"rtmpplaypath\" option, too.\n\nlisten\nAct as a server, listening for an incoming connection.\n\ntimeout\nMaximum time to wait for the incoming connection. Implies listen.\n\nAdditionally, the following parameters can be set via command line options (or in code via\n\"AVOption\"s):\n\nrtmpapp\nName of application to connect on the RTMP server. This option overrides the parameter\nspecified in the URI.\n\nrtmpbuffer\nSet the client buffer time in milliseconds. The default is 3000.\n\nrtmpconn\nExtra arbitrary AMF connection parameters, parsed from a string, e.g. like \"B:1 S:authMe\nO:1 NN:code:1.23 NS:flag:ok O:0\".  Each value is prefixed by a single character denoting\nthe type, B for Boolean, N for number, S for string, O for object, or Z for null,\nfollowed by a colon. For Booleans the data must be either 0 or 1 for FALSE or TRUE,\nrespectively.  Likewise for Objects the data must be 0 or 1 to end or begin an object,\nrespectively. Data items in subobjects may be named, by prefixing the type with 'N' and\nspecifying the name before the value (i.e. \"NB:myFlag:1\"). This option may be used\nmultiple times to construct arbitrary AMF sequences.\n\nrtmpflashver\nVersion of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2.\n(When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)\n\nrtmpflushinterval\nNumber of packets flushed in the same request (RTMPT only). The default is 10.\n\nrtmplive\nSpecify that the media is a live stream. No resuming or seeking in live streams is\npossible. The default value is \"any\", which means the subscriber first tries to play the\nlive stream specified in the playpath. If a live stream of that name is not found, it\nplays the recorded stream. The other possible values are \"live\" and \"recorded\".\n\nrtmppageurl\nURL of the web page in which the media was embedded. By default no value will be sent.\n\nrtmpplaypath\nStream identifier to play or to publish. This option overrides the parameter specified in\nthe URI.\n\nrtmpsubscribe\nName of live stream to subscribe to. By default no value will be sent.  It is only sent\nif the option is specified or if rtmplive is set to live.\n\nrtmpswfhash\nSHA256 hash of the decompressed SWF file (32 bytes).\n\nrtmpswfsize\nSize of the decompressed SWF file, required for SWFVerification.\n\nrtmpswfurl\nURL of the SWF player for the media. By default no value will be sent.\n\nrtmpswfverify\nURL to player swf file, compute hash/size automatically.\n\nrtmptcurl\nURL of the target stream. Defaults to proto://host[:port]/app.\n\nFor example to read with ffplay a multimedia resource named \"sample\" from the application\n\"vod\" from an RTMP server \"myserver\":\n\nffplay rtmp://myserver/vod/sample\n\nTo publish to a password protected server, passing the playpath and app names separately:\n\nffmpeg -re -i <input> -f flv -rtmpplaypath some/long/path -rtmpapp long/app/name rtmp://username:password@myserver/\n\nrtmpe\nEncrypted Real-Time Messaging Protocol.\n\nThe Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia content\nwithin standard cryptographic primitives, consisting of Diffie-Hellman key exchange and\nHMACSHA256, generating a pair of RC4 keys.\n\nrtmps\nReal-Time Messaging Protocol over a secure SSL connection.\n\nThe Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across an\nencrypted connection.\n\nrtmpt\nReal-Time Messaging Protocol tunneled through HTTP.\n\nThe Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming\nmultimedia content within HTTP requests to traverse firewalls.\n\nrtmpte\nEncrypted Real-Time Messaging Protocol tunneled through HTTP.\n\nThe Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for\nstreaming multimedia content within HTTP requests to traverse firewalls.\n\nrtmpts\nReal-Time Messaging Protocol tunneled through HTTPS.\n\nThe Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming\nmultimedia content within HTTPS requests to traverse firewalls.\n\nlibsmbclient\nlibsmbclient permits one to manipulate CIFS/SMB network resources.\n\nFollowing syntax is required.\n\nsmb://[[domain:]user[:password@]]server[/share[/path[/file]]]\n\nThis protocol accepts the following options.\n\ntimeout\nSet timeout in milliseconds of socket I/O operations used by the underlying low level\noperation. By default it is set to -1, which means that the timeout is not specified.\n\ntruncate\nTruncate existing files on write, if set to 1. A value of 0 prevents truncating. Default\nvalue is 1.\n\nworkgroup\nSet the workgroup used for making connections. By default workgroup is not specified.\n\nFor more information see: <http://www.samba.org/>.\n\nlibssh\nSecure File Transfer Protocol via libssh\n\nRead from or write to remote resources using SFTP protocol.\n\nFollowing syntax is required.\n\nsftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg\n\nThis protocol accepts the following options.\n\ntimeout\nSet timeout of socket I/O operations used by the underlying low level operation. By\ndefault it is set to -1, which means that the timeout is not specified.\n\ntruncate\nTruncate existing files on write, if set to 1. A value of 0 prevents truncating. Default\nvalue is 1.\n\nprivatekey\nSpecify the path of the file containing private key to use during authorization.  By\ndefault libssh searches for keys in the ~/.ssh/ directory.\n\nExample: Play a file stored on remote server.\n\nffplay sftp://user:password@serveraddress:22/home/user/resource.mpeg\n\nlibrtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte\nReal-Time Messaging Protocol and its variants supported through librtmp.\n\nRequires the presence of the librtmp headers and library during configuration. You need to\nexplicitly configure the build with \"--enable-librtmp\". If enabled this will replace the\nnative RTMP protocol.\n\nThis protocol provides most client functions and a few server functions needed to support\nRTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and\ntunneled variants of these encrypted types (RTMPTE, RTMPTS).\n\nThe required syntax is:\n\n<rtmpproto>://<server>[:<port>][/<app>][/<playpath>] <options>\n\nwhere rtmpproto is one of the strings \"rtmp\", \"rtmpt\", \"rtmpe\", \"rtmps\", \"rtmpte\", \"rtmpts\"\ncorresponding to each RTMP variant, and server, port, app and playpath have the same meaning\nas specified for the RTMP native protocol.  options contains a list of space-separated\noptions of the form key=val.\n\nSee the librtmp manual page (man 3 librtmp) for more information.\n\nFor example, to stream a file in real-time to an RTMP server using ffmpeg:\n\nffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream\n\nTo play the same stream using ffplay:\n\nffplay \"rtmp://myserver/live/mystream live=1\"\n\nrtp\nReal-time Transport Protocol.\n\nThe required syntax for an RTP URL is: rtp://hostname[:port][?option=val...]\n\nport specifies the RTP port to use.\n\nThe following URL options are supported:\n\nttl=n\nSet the TTL (Time-To-Live) value (for multicast only).\n\nrtcpport=n\nSet the remote RTCP port to n.\n\nlocalrtpport=n\nSet the local RTP port to n.\n\nlocalrtcpport=n'\nSet the local RTCP port to n.\n\npktsize=n\nSet max packet size (in bytes) to n.\n\nbuffersize=size\nSet the maximum UDP socket buffer size in bytes.\n\nconnect=0|1\nDo a \"connect()\" on the UDP socket (if set to 1) or not (if set to 0).\n\nsources=ip[,ip]\nList allowed source IP addresses.\n\nblock=ip[,ip]\nList disallowed (blocked) source IP addresses.\n\nwritetosource=0|1\nSend packets to the source address of the latest received packet (if set to 1) or to a\ndefault remote address (if set to 0).\n\nlocalport=n\nSet the local RTP port to n.\n\ntimeout=n\nSet timeout (in microseconds) of socket I/O operations to n.\n\nThis is a deprecated option. Instead, localrtpport should be used.\n\nImportant notes:\n\n1.  If rtcpport is not set the RTCP port will be set to the RTP port value plus 1.\n\n2.  If localrtpport (the local RTP port) is not set any available port will be used for the\nlocal RTP and RTCP ports.\n\n3.  If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port\nvalue plus 1.\n\nrtsp\nReal-Time Streaming Protocol.\n\nRTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The\ndemuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple\nand Microsoft) and Real-RTSP (with data transferred over RDT).\n\nThe muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it\n(currently Darwin Streaming Server and Mischa Spiegelmock's\n<https://github.com/revmischa/rtsp-server>).\n\nThe required syntax for a RTSP url is:\n\nrtsp://<hostname>[:<port>]/<path>\n\nOptions can be set on the ffmpeg/ffplay command line, or set in code via \"AVOption\"s or in\n\"avformatopeninput\".\n\nThe following options are supported.\n\ninitialpause\nDo not start playing the stream immediately if set to 1. Default value is 0.\n\nrtsptransport\nSet RTSP transport protocols.\n\nIt accepts the following values:\n\nudp Use UDP as lower transport protocol.\n\ntcp Use TCP (interleaving within the RTSP control channel) as lower transport protocol.\n\nudpmulticast\nUse UDP multicast as lower transport protocol.\n\nhttp\nUse HTTP tunneling as lower transport protocol, which is useful for passing proxies.\n\nMultiple lower transport protocols may be specified, in that case they are tried one at a\ntime (if the setup of one fails, the next one is tried).  For the muxer, only the tcp and\nudp options are supported.\n\nrtspflags\nSet RTSP flags.\n\nThe following values are accepted:\n\nfiltersrc\nAccept packets only from negotiated peer address and port.\n\nlisten\nAct as a server, listening for an incoming connection.\n\nprefertcp\nTry TCP for RTP transport first, if TCP is available as RTSP RTP transport.\n\nDefault value is none.\n\nallowedmediatypes\nSet media types to accept from the server.\n\nThe following flags are accepted:\n\nvideo\naudio\ndata\n\nBy default it accepts all media types.\n\nminport\nSet minimum local UDP port. Default value is 5000.\n\nmaxport\nSet maximum local UDP port. Default value is 65000.\n\ntimeout\nSet maximum timeout (in seconds) to wait for incoming connections.\n\nA value of -1 means infinite (default). This option implies the rtspflags set to listen.\n\nreorderqueuesize\nSet number of packets to buffer for handling of reordered packets.\n\nstimeout\nSet socket TCP I/O timeout in microseconds.\n\nuser-agent\nOverride User-Agent header. If not specified, it defaults to the libavformat identifier\nstring.\n\nWhen receiving data over UDP, the demuxer tries to reorder received packets (since they may\narrive out of order, or packets may get lost totally). This can be disabled by setting the\nmaximum demuxing delay to zero (via the \"maxdelay\" field of AVFormatContext).\n\nWhen watching multi-bitrate Real-RTSP streams with ffplay, the streams to display can be\nchosen with \"-vst\" n and \"-ast\" n for video and audio respectively, and can be switched on\nthe fly by pressing \"v\" and \"a\".\n\nExamples\n\nThe following examples all make use of the ffplay and ffmpeg tools.\n\no   Watch a stream over UDP, with a max reordering delay of 0.5 seconds:\n\nffplay -maxdelay 500000 -rtsptransport udp rtsp://server/video.mp4\n\no   Watch a stream tunneled over HTTP:\n\nffplay -rtsptransport http rtsp://server/video.mp4\n\no   Send a stream in realtime to a RTSP server, for others to watch:\n\nffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp\n\no   Receive a stream in realtime:\n\nffmpeg -rtspflags listen -i rtsp://ownaddress/live.sdp <output>\n\nsap\nSession Announcement Protocol (RFC 2974). This is not technically a protocol handler in\nlibavformat, it is a muxer and demuxer.  It is used for signalling of RTP streams, by\nannouncing the SDP for the streams regularly on a separate port.\n\nMuxer\n\nThe syntax for a SAP url given to the muxer is:\n\nsap://<destination>[:<port>][?<options>]\n\nThe RTP packets are sent to destination on port port, or to port 5004 if no port is\nspecified.  options is a \"&\"-separated list. The following options are supported:\n\nannounceaddr=address\nSpecify the destination IP address for sending the announcements to.  If omitted, the\nannouncements are sent to the commonly used SAP announcement multicast address\n224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.\n\nannounceport=port\nSpecify the port to send the announcements on, defaults to 9875 if not specified.\n\nttl=ttl\nSpecify the time to live value for the announcements and RTP packets, defaults to 255.\n\nsameport=0|1\nIf set to 1, send all RTP streams on the same port pair. If zero (the default), all\nstreams are sent on unique ports, with each stream on a port 2 numbers higher than the\nprevious.  VLC/Live555 requires this to be set to 1, to be able to receive the stream.\nThe RTP stack in libavformat for receiving requires all streams to be sent on unique\nports.\n\nExample command lines follow.\n\nTo broadcast a stream on the local subnet, for watching in VLC:\n\nffmpeg -re -i <input> -f sap sap://224.0.0.255?sameport=1\n\nSimilarly, for watching in ffplay:\n\nffmpeg -re -i <input> -f sap sap://224.0.0.255\n\nAnd for watching in ffplay, over IPv6:\n\nffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]\n\nDemuxer\n\nThe syntax for a SAP url given to the demuxer is:\n\nsap://[<address>][:<port>]\n\naddress is the multicast address to listen for announcements on, if omitted, the default\n224.2.127.254 (sap.mcast.net) is used. port is the port that is listened on, 9875 if omitted.\n\nThe demuxers listens for announcements on the given address and port.  Once an announcement\nis received, it tries to receive that particular stream.\n\nExample command lines follow.\n\nTo play back the first stream announced on the normal SAP multicast address:\n\nffplay sap://\n\nTo play back the first stream announced on one the default IPv6 SAP multicast address:\n\nffplay sap://[ff0e::2:7ffe]\n\nsctp\nStream Control Transmission Protocol.\n\nThe accepted URL syntax is:\n\nsctp://<host>:<port>[?<options>]\n\nThe protocol accepts the following options:\n\nlisten\nIf set to any value, listen for an incoming connection. Outgoing connection is done by\ndefault.\n\nmaxstreams\nSet the maximum number of streams. By default no limit is set.\n\nsrt\nHaivision Secure Reliable Transport Protocol via libsrt.\n\nThe supported syntax for a SRT URL is:\n\nsrt://<hostname>:<port>[?<options>]\n\noptions contains a list of &-separated options of the form key=val.\n\nor\n\n<options> srt://<hostname>:<port>\n\noptions contains a list of '-key val' options.\n\nThis protocol accepts the following options.\n\nconnecttimeout=milliseconds\nConnection timeout; SRT cannot connect for RTT > 1500 msec (2 handshake exchanges) with\nthe default connect timeout of 3 seconds. This option applies to the caller and\nrendezvous connection modes. The connect timeout is 10 times the value set for the\nrendezvous mode (which can be used as a workaround for this connection problem with\nearlier versions).\n\nffs=bytes\nFlight Flag Size (Window Size), in bytes. FFS is actually an internal parameter and you\nshould set it to not less than recvbuffersize and mss. The default value is relatively\nlarge, therefore unless you set a very large receiver buffer, you do not need to change\nthis option. Default value is 25600.\n\ninputbw=bytes/seconds\nSender nominal input rate, in bytes per seconds. Used along with oheadbw, when maxbw is\nset to relative (0), to calculate maximum sending rate when recovery packets are sent\nalong with the main media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set\nwhile maxbw is set to relative (0), the actual input rate is evaluated inside the\nlibrary. Default value is 0.\n\niptos=tos\nIP Type of Service. Applies to sender only. Default value is 0xB8.\n\nipttl=ttl\nIP Time To Live. Applies to sender only. Default value is 64.\n\nlatency=microseconds\nTimestamp-based Packet Delivery Delay.  Used to absorb bursts of missed packet\nretransmissions.  This flag sets both rcvlatency and peerlatency to the same value. Note\nthat prior to version 1.3.0 this is the only flag to set the latency, however this is\neffectively equivalent to setting peerlatency, when side is sender and rcvlatency when\nside is receiver, and the bidirectional stream sending is not supported.\n\nlistentimeout=microseconds\nSet socket listen timeout.\n\nmaxbw=bytes/seconds\nMaximum sending bandwidth, in bytes per seconds.  -1 infinite (CSRTCC limit is 30mbps) 0\nrelative to input rate (see inputbw) >0 absolute limit value Default value is 0\n(relative)\n\nmode=caller|listener|rendezvous\nConnection mode.  caller opens client connection.  listener starts server to listen for\nincoming connections.  rendezvous use Rendez-Vous connection mode.  Default value is\ncaller.\n\nmss=bytes\nMaximum Segment Size, in bytes. Used for buffer allocation and rate calculation using a\npacket counter assuming fully filled packets. The smallest MSS between the peers is used.\nThis is 1500 by default in the overall internet.  This is the maximum size of the UDP\npacket and can be only decreased, unless you have some unusual dedicated network\nsettings. Default value is 1500.\n\nnakreport=1|0\nIf set to 1, Receiver will send `UMSGLOSSREPORT` messages periodically until a lost\npacket is retransmitted or intentionally dropped. Default value is 1.\n\noheadbw=percents\nRecovery bandwidth overhead above input rate, in percents.  See inputbw. Default value is\n25%.\n\npassphrase=string\nHaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. The\npassphrase is the shared secret between the sender and the receiver. It is used to\ngenerate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function). It\nis used only if pbkeylen is non-zero. It is used on the receiver only if the received\ndata is encrypted.  The configured passphrase cannot be recovered (write-only).\n\nenforcedencryption=1|0\nIf true, both connection parties must have the same password set (including empty, that\nis, with no encryption). If the password doesn't match or only one side is unencrypted,\nthe connection is rejected. Default is true.\n\nkmrefreshrate=packets\nThe number of packets to be transmitted after which the encryption key is switched to a\nnew key. Default is -1.  -1 means auto (0x1000000 in srt library). The range for this\noption is integers in the 0 - \"INTMAX\".\n\nkmpreannounce=packets\nThe interval between when a new encryption key is sent and when switchover occurs. This\nvalue also applies to the subsequent interval between when switchover occurs and when the\nold encryption key is decommissioned. Default is -1.  -1 means auto (0x1000 in srt\nlibrary). The range for this option is integers in the 0 - \"INTMAX\".\n\npayloadsize=bytes\nSets the maximum declared size of a packet transferred during the single call to the\nsending function in Live mode. Use 0 if this value isn't used (which is default in file\nmode).  Default is -1 (automatic), which typically means MPEG-TS; if you are going to use\nSRT to send any different kind of payload, such as, for example, wrapping a live stream\nin very small frames, then you can use a bigger maximum frame size, though not greater\nthan 1456 bytes.\n\npktsize=bytes\nAlias for payloadsize.\n\npeerlatency=microseconds\nThe latency value (as described in rcvlatency) that is set by the sender side as a\nminimum value for the receiver.\n\npbkeylen=bytes\nSender encryption key length, in bytes.  Only can be set to 0, 16, 24 and 32.  Enable\nsender encryption if not 0.  Not required on receiver (set to 0), key size obtained from\nsender in HaiCrypt handshake.  Default value is 0.\n\nrcvlatency=microseconds\nThe time that should elapse since the moment when the packet was sent and the moment when\nit's delivered to the receiver application in the receiving function.  This time should\nbe a buffer time large enough to cover the time spent for sending, unexpectedly extended\nRTT time, and the time needed to retransmit the lost UDP packet. The effective latency\nvalue will be the maximum of this options' value and the value of peerlatency set by the\npeer side. Before version 1.3.0 this option is only available as latency.\n\nrecvbuffersize=bytes\nSet UDP receive buffer size, expressed in bytes.\n\nsendbuffersize=bytes\nSet UDP send buffer size, expressed in bytes.\n\ntimeout=microseconds\nSet raise error timeouts for read, write and connect operations. Note that the SRT\nlibrary has internal timeouts which can be controlled separately, the value set here is\nonly a cap on those.\n\ntlpktdrop=1|0\nToo-late Packet Drop. When enabled on receiver, it skips missing packets that have not\nbeen delivered in time and delivers the following packets to the application when their\ntime-to-play has come. It also sends a fake ACK to the sender. When enabled on sender and\nenabled on the receiving peer, the sender drops the older packets that have no chance of\nbeing delivered in time. It was automatically enabled in the sender if the receiver\nsupports it.\n\nsndbuf=bytes\nSet send buffer size, expressed in bytes.\n\nrcvbuf=bytes\nSet receive buffer size, expressed in bytes.\n\nReceive buffer must not be greater than ffs.\n\nlossmaxttl=packets\nThe value up to which the Reorder Tolerance may grow. When Reorder Tolerance is > 0, then\npacket loss report is delayed until that number of packets come in. Reorder Tolerance\nincreases every time a \"belated\" packet has come, but it wasn't due to retransmission\n(that is, when UDP packets tend to come out of order), with the difference between the\nlatest sequence and this packet's sequence, and not more than the value of this option.\nBy default it's 0, which means that this mechanism is turned off, and the loss report is\nalways sent immediately upon experiencing a \"gap\" in sequences.\n\nminversion\nThe minimum SRT version that is required from the peer. A connection to a peer that does\nnot satisfy the minimum version requirement will be rejected.\n\nThe version format in hex is 0xXXYYZZ for x.y.z in human readable form.\n\nstreamid=string\nA string limited to 512 characters that can be set on the socket prior to connecting.\nThis stream ID will be able to be retrieved by the listener side from the socket that is\nreturned from srtaccept and was connected by a socket with that set stream ID. SRT does\nnot enforce any special interpretation of the contents of this string.  This option\ndoesnXt make sense in Rendezvous connection; the result might be that simply one side\nwill override the value from the other side and itXs the matter of luck which one would\nwin\n\nsmoother=live|file\nThe type of Smoother used for the transmission for that socket, which is responsible for\nthe transmission and congestion control. The Smoother type must be exactly the same on\nboth connecting parties, otherwise the connection is rejected.\n\nmessageapi=1|0\nWhen set, this socket uses the Message API, otherwise it uses Buffer API. Note that in\nlive mode (see transtype) thereXs only message API available. In File mode you can chose\nto use one of two modes:\n\nStream API (default, when this option is false). In this mode you may send as many data\nas you wish with one sending instruction, or even use dedicated functions that read\ndirectly from a file. The internal facility will take care of any speed and congestion\ncontrol. When receiving, you can also receive as many data as desired, the data not\nextracted will be waiting for the next call. There is no boundary between data portions\nin the Stream mode.\n\nMessage API. In this mode your single sending instruction passes exactly one piece of\ndata that has boundaries (a message). Contrary to Live mode, this message may span across\nmultiple UDP packets and the only size limitation is that it shall fit as a whole in the\nsending buffer. The receiver shall use as large buffer as necessary to receive the\nmessage, otherwise the message will not be given up. When the message is not complete\n(not all packets received or there was a packet loss) it will not be given up.\n\ntranstype=live|file\nSets the transmission type for the socket, in particular, setting this option sets\nmultiple other parameters to their default values as required for a particular\ntransmission type.\n\nlive: Set options as for live transmission. In this mode, you should send by one sending\ninstruction only so many data that fit in one UDP packet, and limited to the value\ndefined first in payloadsize (1316 is default in this mode). There is no speed control\nin this mode, only the bandwidth control, if configured, in order to not exceed the\nbandwidth with the overhead transmission (retransmitted and control packets).\n\nfile: Set options as for non-live transmission. See messageapi for further explanations\n\nlinger=seconds\nThe number of seconds that the socket waits for unsent data when closing.  Default is -1.\n-1 means auto (off with 0 seconds in live mode, on with 180 seconds in file mode). The\nrange for this option is integers in the 0 - \"INTMAX\".\n\nFor more information see: <https://github.com/Haivision/srt>.\n\nsrtp\nSecure Real-time Transport Protocol.\n\nThe accepted options are:\n\nsrtpinsuite\nsrtpoutsuite\nSelect input and output encoding suites.\n\nSupported values:\n\nAESCM128HMACSHA180\nSRTPAES128CMHMACSHA180\nAESCM128HMACSHA132\nSRTPAES128CMHMACSHA132\nsrtpinparams\nsrtpoutparams\nSet input and output encoding parameters, which are expressed by a base64-encoded\nrepresentation of a binary block. The first 16 bytes of this binary block are used as\nmaster key, the following 14 bytes are used as master salt.\n\nsubfile\nVirtually extract a segment of a file or another stream.  The underlying stream must be\nseekable.\n\nAccepted options:\n\nstart\nStart offset of the extracted segment, in bytes.\n\nend End offset of the extracted segment, in bytes.  If set to 0, extract till end of file.\n\nExamples:\n\nExtract a chapter from a DVD VOB file (start and end sectors obtained externally and\nmultiplied by 2048):\n\nsubfile,,start,153391104,end,268142592,,:/media/dvd/VIDEOTS/VTS081.VOB\n\nPlay an AVI file directly from a TAR archive:\n\nsubfile,,start,183241728,end,366490624,,:archive.tar\n\nPlay a MPEG-TS file from start offset till end:\n\nsubfile,,start,32815239,end,0,,:video.ts\n\ntee\nWrites the output to multiple protocols. The individual outputs are separated by |\n\ntee:file://path/to/local/this.avi|file://path/to/local/that.avi\n\ntcp\nTransmission Control Protocol.\n\nThe required syntax for a TCP url is:\n\ntcp://<hostname>:<port>[?<options>]\n\noptions contains a list of &-separated options of the form key=val.\n\nThe list of supported options follows.\n\nlisten=2|1|0\nListen for an incoming connection. 0 disables listen, 1 enables listen in single client\nmode, 2 enables listen in multi-client mode. Default value is 0.\n\ntimeout=microseconds\nSet raise error timeout, expressed in microseconds.\n\nThis option is only relevant in read mode: if no data arrived in more than this time\ninterval, raise error.\n\nlistentimeout=milliseconds\nSet listen timeout, expressed in milliseconds.\n\nrecvbuffersize=bytes\nSet receive buffer size, expressed bytes.\n\nsendbuffersize=bytes\nSet send buffer size, expressed bytes.\n\ntcpnodelay=1|0\nSet TCPNODELAY to disable Nagle's algorithm. Default value is 0.\n\ntcpmss=bytes\nSet maximum segment size for outgoing TCP packets, expressed in bytes.\n\nThe following example shows how to setup a listening TCP connection with ffmpeg, which is\nthen accessed with ffplay:\n\nffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen\nffplay tcp://<hostname>:<port>\n\ntls\nTransport Layer Security (TLS) / Secure Sockets Layer (SSL)\n\nThe required syntax for a TLS/SSL url is:\n\ntls://<hostname>:<port>[?<options>]\n\nThe following parameters can be set via command line options (or in code via \"AVOption\"s):\n\ncafile, cafile=filename\nA file containing certificate authority (CA) root certificates to treat as trusted. If\nthe linked TLS library contains a default this might not need to be specified for\nverification to work, but not all libraries and setups have defaults built in.  The file\nmust be in OpenSSL PEM format.\n\ntlsverify=1|0\nIf enabled, try to verify the peer that we are communicating with.  Note, if using\nOpenSSL, this currently only makes sure that the peer certificate is signed by one of the\nroot certificates in the CA database, but it does not validate that the certificate\nactually matches the host name we are trying to connect to. (With other backends, the\nhost name is validated as well.)\n\nThis is disabled by default since it requires a CA database to be provided by the caller\nin many cases.\n\ncertfile, cert=filename\nA file containing a certificate to use in the handshake with the peer.  (When operating\nas server, in listen mode, this is more often required by the peer, while client\ncertificates only are mandated in certain setups.)\n\nkeyfile, key=filename\nA file containing the private key for the certificate.\n\nlisten=1|0\nIf enabled, listen for connections on the provided port, and assume the server role in\nthe handshake instead of the client role.\n\nhttpproxy\nThe HTTP proxy to tunnel through, e.g. \"http://example.com:1234\".  The proxy must support\nthe CONNECT method.\n\nExample command lines:\n\nTo create a TLS/SSL server that serves an input stream.\n\nffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>\n\nTo play back a stream from the TLS/SSL server using ffplay:\n\nffplay tls://<hostname>:<port>\n\nudp\nUser Datagram Protocol.\n\nThe required syntax for an UDP URL is:\n\nudp://<hostname>:<port>[?<options>]\n\noptions contains a list of &-separated options of the form key=val.\n\nIn case threading is enabled on the system, a circular buffer is used to store the incoming\ndata, which allows one to reduce loss of data due to UDP socket buffer overruns. The\nfifosize and overrunnonfatal options are related to this buffer.\n\nThe list of supported options follows.\n\nbuffersize=size\nSet the UDP maximum socket buffer size in bytes. This is used to set either the receive\nor send buffer size, depending on what the socket is used for.  Default is 32 KB for\noutput, 384 KB for input.  See also fifosize.\n\nbitrate=bitrate\nIf set to nonzero, the output will have the specified constant bitrate if the input has\nenough packets to sustain it.\n\nburstbits=bits\nWhen using bitrate this specifies the maximum number of bits in packet bursts.\n\nlocalport=port\nOverride the local UDP port to bind with.\n\nlocaladdr=addr\nLocal IP address of a network interface used for sending packets or joining multicast\ngroups.\n\npktsize=size\nSet the size in bytes of UDP packets.\n\nreuse=1|0\nExplicitly allow or disallow reusing UDP sockets.\n\nttl=ttl\nSet the time to live value (for multicast only).\n\nconnect=1|0\nInitialize the UDP socket with \"connect()\". In this case, the destination address can't\nbe changed with ffudpsetremoteurl later.  If the destination address isn't known at\nthe start, this option can be specified in ffudpsetremoteurl, too.  This allows\nfinding out the source address for the packets with getsockname, and makes writes return\nwith AVERROR(ECONNREFUSED) if \"destination unreachable\" is received.  For receiving, this\ngives the benefit of only receiving packets from the specified peer address/port.\n\nsources=address[,address]\nOnly receive packets sent from the specified addresses. In case of multicast, also\nsubscribe to multicast traffic coming from these addresses only.\n\nblock=address[,address]\nIgnore packets sent from the specified addresses. In case of multicast, also exclude the\nsource addresses in the multicast subscription.\n\nfifosize=units\nSet the UDP receiving circular buffer size, expressed as a number of packets with size of\n188 bytes. If not specified defaults to 7*4096.\n\noverrunnonfatal=1|0\nSurvive in case of UDP receiving circular buffer overrun. Default value is 0.\n\ntimeout=microseconds\nSet raise error timeout, expressed in microseconds.\n\nThis option is only relevant in read mode: if no data arrived in more than this time\ninterval, raise error.\n\nbroadcast=1|0\nExplicitly allow or disallow UDP broadcasting.\n\nNote that broadcasting may not work properly on networks having a broadcast storm\nprotection.\n\nExamples\n\no   Use ffmpeg to stream over UDP to a remote endpoint:\n\nffmpeg -i <input> -f <format> udp://<hostname>:<port>\n\no   Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP packets, using a large\ninput buffer:\n\nffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pktsize=188&buffersize=65535\n\no   Use ffmpeg to receive over UDP from a remote endpoint:\n\nffmpeg -i udp://[<multicast-address>]:<port> ...\n\nunix\nUnix local socket\n\nThe required syntax for a Unix socket URL is:\n\nunix://<filepath>\n\nThe following parameters can be set via command line options (or in code via \"AVOption\"s):\n\ntimeout\nTimeout in ms.\n\nlisten\nCreate the Unix socket in listening mode.\n\nzmq\nZeroMQ asynchronous messaging using the libzmq library.\n\nThis library supports unicast streaming to multiple clients without relying on an external\nserver.\n\nThe required syntax for streaming or connecting to a stream is:\n\nzmq:tcp://ip-address:port\n\nExample: Create a localhost stream on port 5555:\n\nffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555\n\nMultiple clients may connect to the stream using:\n\nffplay zmq:tcp://127.0.0.1:5555\n\nStreaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern.  The server side\nbinds to a port and publishes data. Clients connect to the server (via IP address/port) and\nsubscribe to the stream. The order in which the server and client start generally does not\nmatter.\n\nffmpeg must be compiled with the --enable-libzmq option to support this protocol.\n\nOptions can be set on the ffmpeg/ffplay command line. The following options are supported:\n\npktsize\nForces the maximum packet size for sending/receiving data. The default value is 131,072\nbytes. On the server side, this sets the maximum size of sent packets via ZeroMQ. On the\nclients, it sets an internal buffer size for receiving packets. Note that pktsize on the\nclients should be equal to or greater than pktsize on the server. Otherwise the received\nmessage may be truncated causing decoding errors.\n",
                "subsections": []
            },
            "SEE ALSO": {
                "content": "ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)\n",
                "subsections": []
            },
            "AUTHORS": {
                "content": "The FFmpeg developers.\n\nFor details about the authorship, see the Git history of the project\n(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in the FFmpeg source\ndirectory, or browsing the online repository at <http://source.ffmpeg.org>.\n\nMaintainers for the specific components are listed in the file MAINTAINERS in the source code\ntree.\n\nFFMPEG-PROTOCOLS(1)",
                "subsections": []
            }
        }
    }
}