This document describes the input and output protocols provided by the +libavformat library. +
+ + +The libavformat library provides some generic global options, which +can be set on all the protocols. In addition each protocol may support +so-called private options, which are specific for that component. +
+Options may be set by specifying -option value in the
+FFmpeg tools, or by setting the value explicitly in the
+AVFormatContext options or using the libavutil/opt.h API
+for programmatic use.
+
The list of supported options follows: +
+Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols +prefixed by "-" are disabled. +All protocols are allowed by default but protocols used by an another +protocol (nested protocols) are restricted to a per protocol subset. +
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=PROTOCOL", or you can disable a +particular protocol using the option +"–disable-protocol=PROTOCOL". +
+The option "-protocols" of the ff* tools will display the list of +supported protocols. +
+All protocols accept the following options: +
+Maximum time to wait for (network) read/write operations to complete, +in microseconds. +
A description of the currently available protocols follows. +
+ +Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based +publish-subscribe communication protocol. +
+FFmpeg must be compiled with –enable-librabbitmq to support AMQP. A separate +AMQP broker must also be run. An example open-source AMQP broker is RabbitMQ. +
+After starting the broker, an FFmpeg client may stream data to the broker using +the command: +
+ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost] +
Where hostname and port (default is 5672) is the address of the broker. The +client may also set a user/password for authentication. The default for both +fields is "guest". Name of virtual host on broker can be set with vhost. The +default value is "/". +
+Muliple subscribers may stream from the broker using the command: +
ffplay amqp://[[user]:[password]@]hostname[:port][/vhost] +
In RabbitMQ all data published to the broker flows through a specific exchange, +and each subscribing client has an assigned queue/buffer. When a packet arrives +at an exchange, it may be copied to a client’s queue depending on the exchange +and routing_key fields. +
+The following options are supported: +
+Sets the exchange to use on the broker. RabbitMQ has several predefined +exchanges: "amq.direct" is the default exchange, where the publisher and +subscriber must have a matching routing_key; "amq.fanout" is the same as a +broadcast operation (i.e. the data is forwarded to all queues on the fanout +exchange independent of the routing_key); and "amq.topic" is similar to +"amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ +documentation). +
+Sets the routing key. The default value is "amqp". The routing key is used on +the "amq.direct" and "amq.topic" exchanges to decide whether packets are written +to the queue of a subscriber. +
+Maximum size of each packet sent/received to the broker. Default is 131072. +Minimum is 4096 and max is any large value (representable by an int). When +receiving packets, this sets an internal buffer size in FFmpeg. It should be +equal to or greater than the size of the published packets to the broker. Otherwise +the received message may be truncated causing decoding errors. +
+The timeout in seconds during the initial connection to the broker. The +default value is rw_timeout, or 5 seconds if rw_timeout is not set. +
+Sets the delivery mode of each message sent to broker. +The following values are accepted: +
Delivery mode set to "persistent" (2). This is the default value. +Messages may be written to the broker’s disk depending on its setup. +
+Delivery mode set to "non-persistent" (1). +Messages will stay in broker’s memory unless the broker is under memory +pressure. +
+Asynchronous data filling wrapper for input stream. +
+Fill data in a background thread, to decouple I/O operation from demux thread. +
+async:URL +async:http://host/resource +async:cache:http://host/resource +
Read BluRay playlist. +
+The accepted options are: +
BluRay angle +
+Start chapter (1...N) +
+Playlist to read (BDMV/PLAYLIST/?????.mpls) +
+Examples: +
+Read longest playlist from BluRay mounted to /mnt/bluray: +
bluray:/mnt/bluray +
Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: +
-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray +
Caching wrapper for input stream. +
+Cache the input stream to temporary file. It brings seeking capability to live streams. +
+The accepted options are: +
Amount in bytes that may be read ahead when seeking isn’t supported. Range is -1 to INT_MAX. +-1 for unlimited. Default is 65536. +
+URL Syntax is +
cache:URL +
Physical concatenation protocol. +
+Read and seek from many resources in sequence as if they were +a unique resource. +
+A URL accepted by this protocol has the syntax: +
concat:URL1|URL2|...|URLN +
where URL1, URL2, ..., 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 split1.mpeg,
+split2.mpeg, split3.mpeg with ffplay use the
+command:
+
ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg +
Note that you may need to escape the character "|" which is special for +many shells. +
+ +Physical concatenation protocol using a line break delimited list of +resources. +
+Read and seek from many resources in sequence as if they were +a unique resource. +
+A URL accepted by this protocol has the syntax: +
concatf:URL +
where URL is the url containing a line break delimited list of +resources to be concatenated, each one possibly specifying a distinct +protocol. Special characters must be escaped with backslash or single +quotes. See (ffmpeg-utils)the "Quoting and escaping" +section in the ffmpeg-utils(1) manual. +
+For example to read a sequence of files split1.mpeg,
+split2.mpeg, split3.mpeg listed in separate lines within
+a file split.txt with ffplay use the command:
+
ffplay concatf:split.txt +
Where split.txt contains the lines: +
split1.mpeg +split2.mpeg +split3.mpeg +
AES-encrypted stream reading protocol. +
+The accepted options are: +
Set the AES decryption key binary block from given hexadecimal representation. +
+Set the AES decryption initialization vector binary block from given hexadecimal representation. +
Accepted URL formats: +
crypto:URL +crypto+URL +
Data in-line in the URI. See http://en.wikipedia.org/wiki/Data_URI_scheme. +
+For example, to convert a GIF file given inline with ffmpeg:
+
ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png +
File access protocol. +
+Read from or write to a file. +
+A file URL can have the form: +
file:filename +
where filename is the path of the file to read. +
+An URL that does not have a protocol prefix will be assumed to be a +file URL. Depending on the build, an URL that looks like a Windows +path with the drive letter at the beginning will also be assumed to be +a file URL (usually not the case in builds for unix-like systems). +
+For example to read from a file input.mpeg with ffmpeg
+use the command:
+
ffmpeg -i file:input.mpeg output.mpeg +
This protocol accepts the following options: +
+Truncate existing files on write, if set to 1. A value of 0 prevents +truncating. Default value is 1. +
+Set I/O operation maximum block size, in bytes. Default value is
+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.
+
If set to 1, the protocol will retry reading at the end of the file, allowing +reading files that still are being written. In order for this to terminate, +you either need to use the rw_timeout option, or use the interrupt callback +(for API users). +
+Controls if seekability is advertised on the file. 0 means non-seekable, -1 +means auto (seekable for normal files, non-seekable for named pipes). +
+Many demuxers handle seekable and non-seekable resources differently, +overriding this might speed up opening certain files at the cost of losing some +features (e.g. accurate seeking). +
FTP (File Transfer Protocol). +
+Read from or write to remote resources using FTP protocol. +
+Following syntax is required. +
ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg +
This protocol accepts the following options. +
+Set timeout in microseconds 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. +
+Set a user to be used for authenticating to the FTP server. This is overridden by the +user in the FTP URL. +
+Set a password to be used for authenticating to the FTP server. This is overridden by +the password in the FTP URL, or by ftp-anonymous-password if no user is set. +
+Password used when login as anonymous user. Typically an e-mail address +should be used. +
+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. +
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. +
+ +Gopher protocol. +
+ +Gophers protocol. +
+The Gopher protocol with TLS encapsulation. +
+ +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 +"+proto" after the hls URI scheme name, where proto +is either "file" or "http". +
+hls+http://host/path/to/remote/resource.m3u8 +hls+file://path/to/local/resource.m3u8 +
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. +
+ +HTTP (Hyper Text Transfer Protocol). +
+This protocol accepts the following options: +
+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. +
+If set to 1 use chunked Transfer-Encoding for posts, default is 1. +
+Set a specific content type for the POST messages or for listen mode. +
+set HTTP proxy to tunnel through e.g. http://example.com:1234 +
+Set custom HTTP headers, can override built in default headers. The +value must be a string encoding the headers. +
+Use persistent connections if set to 1, default is 0. +
+Set custom HTTP post data. +
+Set the Referer header. Include ’Referer: URL’ header in HTTP request. +
+Override the User-Agent header. If not specified the protocol will use a +string describing the libavformat build. ("Lavf/<version>") +
+If set then eof is treated like an error and causes reconnection, this is useful +for live / endless streams. +
+If set then even streamed/non seekable streams will be reconnected on errors. +
+Reconnect automatically in case of TCP/TLS errors during connect. +
+A comma separated list of HTTP status codes to reconnect on. The list can +include specific status codes (e.g. ’503’) or the strings ’4xx’ / ’5xx’. +
+Sets the maximum delay in seconds after which to give up reconnecting +
+Export the MIME type. +
+Exports the HTTP response version number. Usually "1.0" or "1.1". +
+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 icy_metadata_headers and icy_metadata_packet options. +The default is 1. +
+If the server supports ICY metadata, this contains the ICY-specific HTTP reply +headers, separated by newline characters. +
+If the server supports ICY metadata, and icy was set to 1, this +contains the last non-empty metadata packet sent by the server. It should be +polled in regular intervals by applications interested in mid-stream metadata +updates. +
+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. +
+Set initial byte offset. +
+Try to limit the request to bytes preceding this offset. +
+When used as a client option it sets the HTTP method for the request. +
+When used as a server option it sets the HTTP method that is going to be +expected from the client(s). +If the expected and the received HTTP method do not match the client will +be given a Bad Request response. +When unset the HTTP method is not checked for now. This will be replaced by +autodetection in the future. +
+If set to 1 enables experimental HTTP server. This can be used to send data when +used as an output option, or read data from a client with HTTP POST when used as +an input option. +If set to 2 enables experimental multi-client HTTP server. This is not yet implemented +in ffmpeg.c and thus must not be used as a command line option. +
# Server side (sending): +ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://server:port + +# Client side (receiving): +ffmpeg -i http://server:port -c copy somefile.ogg + +# Client can also be done with wget: +wget http://server:port -O somefile.ogg + +# Server side (receiving): +ffmpeg -listen 1 -i http://server:port -c copy somefile.ogg + +# Client side (sending): +ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://server:port + +# Client can also be done with wget: +wget --post-file=somefile.ogg http://server:port +
Send an Expect: 100-continue header for POST. If set to 1 it will send, if set +to 0 it won’t, if set to -1 it will try to send if it is applicable. Default +value is -1. +
+Set HTTP authentication type. No option for Digest, since this method requires +getting nonce parameters from the server first and can’t be used straight away like +Basic. +
+Choose the HTTP authentication type automatically. This is the default. +
Choose the HTTP basic authentication. +
+Basic authentication sends a Base64-encoded string that contains a user name and password +for the client. Base64 is not a form of encryption and should be considered the same as +sending the user name and password in clear text (Base64 is a reversible encoding). +If a resource needs to be protected, strongly consider using an authentication scheme +other than basic authentication. HTTPS/TLS should be used with basic authentication. +Without these additional security enhancements, basic authentication should not be used +to protect sensitive or valuable information. +
Some HTTP requests will be denied unless cookie values are passed in with the +request. The 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: +
ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8 +
Icecast protocol (stream to Icecast servers) +
+This protocol accepts the following options: +
+Set the stream genre. +
+Set the stream name. +
+Set the stream description. +
+Set the stream website URL. +
+Set if the stream should be public. +The default is 0 (not public). +
+Override the User-Agent header. If not specified a string of the form +"Lavf/<version>" will be used. +
+Set the Icecast mountpoint password. +
+Set the stream content type. This must be set if it is different from +audio/mpeg. +
+This enables support for Icecast versions < 2.4.0, that do not support the +HTTP PUT method but the SOURCE method. +
+Establish a TLS (HTTPS) connection to Icecast. +
+icecast://[username[:password]@]server:port/mountpoint +
InterPlanetary File System (IPFS) protocol support. One can access files stored +on the IPFS network through so-called gateways. These are http(s) endpoints. +This protocol wraps the IPFS native protocols (ipfs:// and ipns://) to be sent +to such a gateway. Users can (and should) host their own node which means this +protocol will use one’s local gateway to access files on the IPFS network. +
+If a user doesn’t have a node of their own then the public gateway https://dweb.link
+is used by default.
+
This protocol accepts the following options: +
+Defines the gateway to use. When not set, the protocol will first try
+locating the local gateway by looking at $IPFS_GATEWAY, $IPFS_PATH
+and $HOME/.ipfs/, in that order. If that fails https://dweb.link will be used.
+
One can use this protocol in 2 ways. Using IPFS: +
ffplay ipfs://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T +
Or the IPNS protocol (IPNS is mutable IPFS): +
ffplay ipns://QmbGtJg23skhvFmu9mJiePVByhfzu5rwo74MEkVDYAmF5T +
MMS (Microsoft Media Server) protocol over TCP. +
+ +MMS (Microsoft Media Server) protocol over HTTP. +
+The required syntax is: +
mmsh://server[:port][/app][/playpath] +
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. +
# 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: +
Note that some formats (typically MOV) require the output protocol to +be seekable, so they will fail with the MD5 output protocol. +
+ +UNIX pipe access protocol. +
+Read and write from UNIX pipes. +
+The accepted syntax is: +
pipe:[number] +
number is the number corresponding to the file descriptor of the +pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If 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 ffmpeg:
+
cat test.wav | ffmpeg -i pipe:0 +# ...this is the same as... +cat test.wav | ffmpeg -i pipe: +
For writing to stdout with ffmpeg:
+
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 +
This protocol accepts the following options: +
+Set I/O operation maximum block size, in bytes. Default value is
+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.
+
Note that some formats (typically MOV), require the output protocol to +be seekable, so they will fail with the pipe output protocol. +
+ +Pro-MPEG Code of Practice #3 Release 2 FEC protocol. +
+The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism +for MPEG-2 Transport Streams sent over RTP. +
+This protocol must be used in conjunction with the rtp_mpegts muxer and
+the rtp protocol.
+
The required syntax is: +
-f rtp_mpegts -fec prompeg=option=val... rtp://hostname:port +
The destination UDP ports are port + 2 for the column FEC stream
+and port + 4 for the row FEC stream.
+
This protocol accepts the following options: +
The number of columns (4-20, LxD <= 100) +
+The number of rows (4-20, LxD <= 100) +
+Example usage: +
+-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://hostname:port +
Reliable Internet Streaming Transport protocol +
+The accepted options are: +
Supported values: +
This one is default. +
Set internal RIST buffer size in milliseconds for retransmission of data. +Default value is 0 which means the librist default (1 sec). Maximum value is 30 +seconds. +
+Size of the librist receiver output fifo in number of packets. This must be a +power of 2. +Defaults to 8192 (vs the librist default of 1024). +
+Survive in case of librist fifo buffer overrun. Default value is 0. +
+Set maximum packet size for sending data. 1316 by default. +
+Set loglevel for RIST logging messages. You only need to set this if you +explicitly want to enable debug level messages or packet loss simulation, +otherwise the regular loglevel is respected. +
+Set override of encryption secret, by default is unset. +
+Set encryption type, by default is disabled. +Acceptable values are 128 and 256. +
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: +
rtmp://[username:password@]server[:port][/app][/instance][/playpath] +
The accepted parameters are: +
An optional username (mostly for publishing). +
+An optional password (mostly for publishing). +
+The address of the RTMP server. +
+The number of the TCP port to use (by default is 1935). +
+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. /ondemand/, /flash/live/, etc.). You can override
+the value parsed from the URI through the rtmp_app option, too.
+
It is the path or name of the resource to play with reference to the
+application specified in app, may be prefixed by "mp4:". You
+can override the value parsed from the URI through the rtmp_playpath
+option, too.
+
Act as a server, listening for an incoming connection. +
+Maximum time to wait for the incoming connection. Implies listen. +
Additionally, the following parameters can be set via command line options
+(or in code via AVOptions):
+
Name of application to connect on the RTMP server. This option +overrides the parameter specified in the URI. +
+Set the client buffer time in milliseconds. The default is 3000. +
+Extra arbitrary AMF connection parameters, parsed from a string,
+e.g. like 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. NB:myFlag:1). This option may be used multiple
+times to construct arbitrary AMF sequences.
+
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>).) +
+Number of packets flushed in the same request (RTMPT only). The default +is 10. +
+Specify that the media is a live stream. No resuming or seeking in
+live streams is possible. The default value is 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 live and
+recorded.
+
URL of the web page in which the media was embedded. By default no +value will be sent. +
+Stream identifier to play or to publish. This option overrides the +parameter specified in the URI. +
+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. +
+SHA256 hash of the decompressed SWF file (32 bytes). +
+Size of the decompressed SWF file, required for SWFVerification. +
+URL of the SWF player for the media. By default no value will be sent. +
+URL to player swf file, compute hash/size automatically. +
+URL of the target stream. Defaults to proto://host[:port]/app. +
+Set TCP_NODELAY to disable Nagle’s algorithm. Default value is 0. +
+Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY. +
+For example to read with ffplay a multimedia resource named
+"sample" from the application "vod" from an RTMP server "myserver":
+
ffplay rtmp://myserver/vod/sample +
To publish to a password protected server, passing the playpath and +app names separately: +
ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/ +
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. +
+ +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. +
+ +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. +
+ +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. +
+ +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. +
+ +libsmbclient permits one to manipulate CIFS/SMB network resources. +
+Following syntax is required. +
+smb://[[domain:]user[:password@]]server[/share[/path[/file]]] +
This protocol accepts the following options. +
+Set timeout in milliseconds of socket I/O operations used by the underlying +low level operation. By default it is set to -1, which means that the timeout +is not specified. +
+Truncate existing files on write, if set to 1. A value of 0 prevents +truncating. Default value is 1. +
+Set the workgroup used for making connections. By default workgroup is not specified. +
+For more information see: http://www.samba.org/. +
+ +Secure File Transfer Protocol via libssh +
+Read from or write to remote resources using SFTP protocol. +
+Following syntax is required. +
+sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg +
This protocol accepts the following options. +
+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. +
+Truncate existing files on write, if set to 1. A value of 0 prevents +truncating. Default value is 1. +
+Specify the path of the file containing private key to use during authorization. +By default libssh searches for keys in the ~/.ssh/ directory. +
+Example: Play a file stored on remote server. +
+ffplay sftp://user:password@server_address:22/home/user/resource.mpeg +
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: +
rtmp_proto://server[:port][/app][/playpath] options +
where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", +"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and +server, port, app and playpath have the same +meaning as specified for the RTMP native protocol. +options contains a list of space-separated options of the form +key=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
+ffmpeg:
+
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream +
To play the same stream using ffplay:
+
ffplay "rtmp://myserver/live/mystream live=1" +
Real-time Transport Protocol. +
+The required syntax for an RTP URL is: +rtp://hostname[:port][?option=val...] +
+port specifies the RTP port to use. +
+The following URL options are supported: +
+Set the TTL (Time-To-Live) value (for multicast only). +
+Set the remote RTCP port to n. +
+Set the local RTP port to n. +
+Set the local RTCP port to n. +
+Set max packet size (in bytes) to n. +
+Set the maximum UDP socket buffer size in bytes. +
+Do a connect() on the UDP socket (if set to 1) or not (if set
+to 0).
+
List allowed source IP addresses. +
+List disallowed (blocked) source IP addresses. +
+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). +
+Set the local RTP port to n. +
+Local IP address of a network interface used for sending packets or joining +multicast groups. +
+Set timeout (in microseconds) of socket I/O operations to n. +
+This is a deprecated option. Instead, localrtpport should be +used. +
+Important notes: +
+Real-Time Streaming Protocol. +
+RTSP is not technically a protocol handler in libavformat, it is a demuxer +and muxer. The demuxer supports both normal RTSP (with data transferred +over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with +data transferred over RDT). +
+The muxer can be used to send a stream using RTSP ANNOUNCE to a server +supporting it (currently Darwin Streaming Server and Mischa Spiegelmock’s +RTSP server). +
+The required syntax for a RTSP url is: +
rtsp://hostname[:port]/path +
Options can be set on the ffmpeg/ffplay command
+line, or set in code via AVOptions or in
+avformat_open_input.
+
The following options are supported. +
+Set RTSP transport protocols. +
+It accepts the following values: +
Use UDP as lower transport protocol. +
+Use TCP (interleaving within the RTSP control channel) as lower +transport protocol. +
Default value is ‘0’. +
+Set RTSP flags. +
+The following values are accepted: +
Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC. +
Use RFC 2190 packetization instead of RFC 4629 for H.263. +
Don’t send RTCP sender reports. +
Use mode 0 for H.264 in RTP. +
Send RTCP BYE packets when finishing. +
Default value is ‘0’. +
+ +Set minimum local UDP port. Default value is 5000. +
+Set maximum local UDP port. Default value is 65000. +
+Set the maximum socket buffer size in bytes. +
+Set max send packet size (in bytes). Default value is 1472. +
The following options are supported. +
+Do not start playing the stream immediately if set to 1. Default value +is 0. +
+Set RTSP transport protocols. +
+It accepts the following values: +
Use UDP as lower transport protocol. +
+Use TCP (interleaving within the RTSP control channel) as lower +transport protocol. +
+Use UDP multicast as lower transport protocol. +
+Use HTTP tunneling as lower transport protocol, which is useful for +passing proxies. +
+Use HTTPs tunneling as lower transport protocol, which is useful for +passing proxies and widely used for security consideration. +
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 ‘tcp’ and ‘udp’ options are supported. +
+Set RTSP flags. +
+The following values are accepted: +
Accept packets only from negotiated peer address and port. +
Act as a server, listening for an incoming connection. +
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport. +
Export raw MPEG-TS stream instead of demuxing. The flag will simply write out +the raw stream, with the original PAT/PMT/PIDs intact. +
Default value is ‘none’. +
+Set media types to accept from the server. +
+The following flags are accepted: +
By default it accepts all media types. +
+Set minimum local UDP port. Default value is 5000. +
+Set maximum local UDP port. Default value is 65000. +
+Set maximum timeout (in seconds) to establish an initial connection. Setting +listen_timeout > 0 sets rtsp_flags to ‘listen’. Default is -1 +which means an infinite timeout when ‘listen’ mode is set. +
+Set number of packets to buffer for handling of reordered packets. +
+Set socket TCP I/O timeout in microseconds. +
+Override User-Agent header. If not specified, it defaults to the +libavformat identifier string. +
+Set the maximum socket buffer size in bytes. +
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 max_delay field of AVFormatContext).
+
When watching multi-bitrate Real-RTSP streams with ffplay, the
+streams to display can be chosen with -vst n and
+-ast n for video and audio respectively, and can be switched
+on the fly by pressing v and a.
+
The following examples all make use of the ffplay and
+ffmpeg tools.
+
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 +
ffplay -rtsp_transport http rtsp://server/video.mp4 +
ffmpeg -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp +
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp output +
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. +
+ +The syntax for a SAP url given to the muxer is: +
sap://destination[:port][?options] +
The RTP packets are sent to destination on port port,
+or to port 5004 if no port is specified.
+options is a &-separated list. The following options
+are supported:
+
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 destination is an IPv6 address. +
+Specify the port to send the announcements on, defaults to +9875 if not specified. +
+Specify the time to live value for the announcements and RTP packets, +defaults to 255. +
+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. +
Example command lines follow. +
+To broadcast a stream on the local subnet, for watching in VLC: +
+ffmpeg -re -i input -f sap sap://224.0.0.255?same_port=1 +
Similarly, for watching in ffplay:
+
ffmpeg -re -i input -f sap sap://224.0.0.255 +
And for watching in ffplay, over IPv6:
+
ffmpeg -re -i input -f sap sap://[ff0e::1:2:3:4] +
The syntax for a SAP url given to the demuxer is: +
sap://[address][:port] +
address is the multicast address to listen for announcements on, +if omitted, the default 224.2.127.254 (sap.mcast.net) is used. 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: +
+ffplay sap:// +
To play back the first stream announced on one the default IPv6 SAP multicast address: +
+ffplay sap://[ff0e::2:7ffe] +
Stream Control Transmission Protocol. +
+The accepted URL syntax is: +
sctp://host:port[?options] +
The protocol accepts the following options: +
If set to any value, listen for an incoming connection. Outgoing connection is done by default. +
+Set the maximum number of streams. By default no limit is set. +
Haivision Secure Reliable Transport Protocol via libsrt. +
+The supported syntax for a SRT URL is: +
srt://hostname:port[?options] +
options contains a list of &-separated options of the form +key=val. +
+or +
+options srt://hostname:port +
options contains a list of ’-key val’ +options. +
+This protocol accepts the following options. +
+Connection timeout; SRT cannot connect for RTT > 1500 msec +(2 handshake exchanges) with the default connect timeout of +3 seconds. This option applies to the caller and rendezvous +connection modes. The connect timeout is 10 times the value +set for the rendezvous mode (which can be used as a +workaround for this connection problem with earlier versions). +
+Flight Flag Size (Window Size), in bytes. FFS is actually an +internal parameter and you should set it to not less than +recv_buffer_size and mss. The default value +is relatively large, therefore unless you set a very large receiver buffer, +you do not need to change this option. Default value is 25600. +
+Sender nominal input rate, in bytes per seconds. Used along with +oheadbw, when maxbw is set to relative (0), to +calculate maximum sending rate when recovery packets are sent +along with the main media stream: +inputbw * (100 + oheadbw) / 100 +if inputbw is not set while maxbw is set to +relative (0), the actual input rate is evaluated inside +the library. Default value is 0. +
+IP Type of Service. Applies to sender only. Default value is 0xB8. +
+IP Time To Live. Applies to sender only. Default value is 64. +
+Timestamp-based Packet Delivery Delay. +Used to absorb bursts of missed packet retransmissions. +This flag sets both rcvlatency and peerlatency +to the same value. Note that prior to version 1.3.0 +this is the only flag to set the latency, however +this is effectively equivalent to setting peerlatency, +when side is sender and rcvlatency +when side is receiver, and the bidirectional stream +sending is not supported. +
+Set socket listen timeout. +
+Maximum sending bandwidth, in bytes per seconds. +-1 infinite (CSRTCC limit is 30mbps) +0 relative to input rate (see inputbw) +>0 absolute limit value +Default value is 0 (relative) +
+Connection mode. +caller opens client connection. +listener starts server to listen for incoming connections. +rendezvous use Rendez-Vous connection mode. +Default value is caller. +
+Maximum Segment Size, in bytes. Used for buffer allocation +and rate calculation using a packet counter assuming fully +filled packets. The smallest MSS between the peers is +used. This is 1500 by default in the overall internet. +This is the maximum size of the UDP packet and can be +only decreased, unless you have some unusual dedicated +network settings. Default value is 1500. +
+If set to 1, Receiver will send ‘UMSG_LOSSREPORT‘ messages +periodically until a lost packet is retransmitted or +intentionally dropped. Default value is 1. +
+Recovery bandwidth overhead above input rate, in percents. +See inputbw. Default value is 25%. +
+HaiCrypt Encryption/Decryption Passphrase string, length +from 10 to 79 characters. The passphrase is the shared +secret between the sender and the receiver. It is used +to generate the Key Encrypting Key using PBKDF2 +(Password-Based Key Derivation Function). It is used +only if pbkeylen is non-zero. It is used on +the receiver only if the received data is encrypted. +The configured passphrase cannot be recovered (write-only). +
+If true, both connection parties must have the same password +set (including empty, that is, with no encryption). If the +password doesn’t match or only one side is unencrypted, +the connection is rejected. Default is true. +
+The number of packets to be transmitted after which the
+encryption key is switched to a new key. Default is -1.
+-1 means auto (0x1000000 in srt library). The range for
+this option is integers in the 0 - INT_MAX.
+
The interval between when a new encryption key is sent and
+when switchover occurs. This value also applies to the
+subsequent interval between when switchover occurs and
+when the old encryption key is decommissioned. Default is -1.
+-1 means auto (0x1000 in srt library). The range for
+this option is integers in the 0 - INT_MAX.
+
The sender’s extra delay before dropping packets. This delay is +added to the default drop delay time interval value. +
+Special value -1: Do not drop packets on the sender at all. +
+Sets the maximum declared size of a packet transferred +during the single call to the sending function in Live +mode. Use 0 if this value isn’t used (which is default in +file mode). +Default is -1 (automatic), which typically means MPEG-TS; +if you are going to use SRT +to send any different kind of payload, such as, for example, +wrapping a live stream in very small frames, then you can +use a bigger maximum frame size, though not greater than +1456 bytes. +
+Alias for ‘payload_size’. +
+The latency value (as described in rcvlatency) that is +set by the sender side as a minimum value for the receiver. +
+Sender encryption key length, in bytes. +Only can be set to 0, 16, 24 and 32. +Enable sender encryption if not 0. +Not required on receiver (set to 0), +key size obtained from sender in HaiCrypt handshake. +Default value is 0. +
+The time that should elapse since the moment when the +packet was sent and the moment when it’s delivered to +the receiver application in the receiving function. +This time should be a buffer time large enough to cover +the time spent for sending, unexpectedly extended RTT +time, and the time needed to retransmit the lost UDP +packet. The effective latency value will be the maximum +of this options’ value and the value of peerlatency +set by the peer side. Before version 1.3.0 this option +is only available as latency. +
+Set UDP receive buffer size, expressed in bytes. +
+Set UDP send buffer size, expressed in bytes. +
+Set raise error timeouts for read, write and connect operations. Note that the +SRT library has internal timeouts which can be controlled separately, the +value set here is only a cap on those. +
+Too-late Packet Drop. When enabled on receiver, it skips +missing packets that have not been delivered in time and +delivers the following packets to the application when +their time-to-play has come. It also sends a fake ACK to +the sender. When enabled on sender and enabled on the +receiving peer, the sender drops the older packets that +have no chance of being delivered in time. It was +automatically enabled in the sender if the receiver +supports it. +
+Set send buffer size, expressed in bytes. +
+Set receive buffer size, expressed in bytes. +
+Receive buffer must not be greater than ffs. +
+The value up to which the Reorder Tolerance may grow. When +Reorder Tolerance is > 0, then packet loss report is delayed +until that number of packets come in. Reorder Tolerance +increases every time a "belated" packet has come, but it +wasn’t due to retransmission (that is, when UDP packets tend +to come out of order), with the difference between the latest +sequence and this packet’s sequence, and not more than the +value of this option. By default it’s 0, which means that this +mechanism is turned off, and the loss report is always sent +immediately upon experiencing a "gap" in sequences. +
+The minimum SRT version that is required from the peer. A connection +to a peer that does not satisfy the minimum version requirement +will be rejected. +
+The version format in hex is 0xXXYYZZ for x.y.z in human readable +form. +
+A string limited to 512 characters that can be set on the socket prior +to connecting. This stream ID will be able to be retrieved by the +listener side from the socket that is returned from srt_accept and +was connected by a socket with that set stream ID. SRT does not enforce +any special interpretation of the contents of this string. +This option doesn’t make sense in Rendezvous connection; the result +might be that simply one side will override the value from the other +side and it’s the matter of luck which one would win +
+Alias for ‘streamid’ to avoid conflict with ffmpeg command line option. +
+The type of Smoother used for the transmission for that socket, which +is responsible for the transmission and congestion control. The Smoother +type must be exactly the same on both connecting parties, otherwise +the connection is rejected. +
+When set, this socket uses the Message API, otherwise it uses Buffer +API. Note that in live mode (see transtype) there’s only +message API available. In File mode you can chose to use one of two modes: +
+Stream API (default, when this option is false). In this mode you may +send as many data as you wish with one sending instruction, or even use +dedicated functions that read directly from a file. The internal facility +will take care of any speed and congestion control. When receiving, you +can also receive as many data as desired, the data not extracted will be +waiting for the next call. There is no boundary between data portions in +the Stream mode. +
+Message API. In this mode your single sending instruction passes exactly +one piece of data that has boundaries (a message). Contrary to Live mode, +this message may span across multiple UDP packets and the only size +limitation is that it shall fit as a whole in the sending buffer. The +receiver shall use as large buffer as necessary to receive the message, +otherwise the message will not be given up. When the message is not +complete (not all packets received or there was a packet loss) it will +not be given up. +
+Sets the transmission type for the socket, in particular, setting this +option sets multiple other parameters to their default values as required +for a particular transmission type. +
+live: Set options as for live transmission. In this mode, you should +send by one sending instruction only so many data that fit in one UDP packet, +and limited to the value defined first in payload_size (1316 is +default in this mode). There is no speed control in this mode, only the +bandwidth control, if configured, in order to not exceed the bandwidth with +the overhead transmission (retransmitted and control packets). +
+file: Set options as for non-live transmission. See messageapi +for further explanations +
+The number of seconds that the socket waits for unsent data when closing.
+Default is -1. -1 means auto (off with 0 seconds in live mode, on with 180
+seconds in file mode). The range for this option is integers in the
+0 - INT_MAX.
+
When true, use Timestamp-based Packet Delivery mode. The default behavior +depends on the transmission type: enabled in live mode, disabled in file +mode. +
+For more information see: https://github.com/Haivision/srt. +
+ +Secure Real-time Transport Protocol. +
+The accepted options are: +
Select input and output encoding suites. +
+Supported values: +
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. +
Virtually extract a segment of a file or another stream. +The underlying stream must be seekable. +
+Accepted options: +
Start offset of the extracted segment, in bytes. +
End offset of the extracted segment, in bytes. +If set to 0, extract till end of file. +
Examples: +
+Extract a chapter from a DVD VOB file (start and end sectors obtained +externally and multiplied by 2048): +
subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB +
Play an AVI file directly from a TAR archive: +
subfile,,start,183241728,end,366490624,,:archive.tar +
Play a MPEG-TS file from start offset till end: +
subfile,,start,32815239,end,0,,:video.ts +
Writes the output to multiple protocols. The individual outputs are separated +by | +
+tee:file://path/to/local/this.avi|file://path/to/local/that.avi +
Transmission Control Protocol. +
+The required syntax for a TCP url is: +
tcp://hostname:port[?options] +
options contains a list of &-separated options of the form +key=val. +
+The list of supported options follows. +
+Listen for an incoming connection. 0 disables listen, 1 enables listen in +single client mode, 2 enables listen in multi-client mode. Default value is 0. +
+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. +
+Set listen timeout, expressed in milliseconds. +
+Set receive buffer size, expressed bytes. +
+Set send buffer size, expressed bytes. +
+Set TCP_NODELAY to disable Nagle’s algorithm. Default value is 0. +
+Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY. +
+Set maximum segment size for outgoing TCP packets, expressed in bytes. +
The following example shows how to setup a listening TCP connection
+with ffmpeg, which is then accessed with ffplay:
+
ffmpeg -i input -f format tcp://hostname:port?listen +ffplay tcp://hostname:port +
Transport Layer Security (TLS) / Secure Sockets Layer (SSL) +
+The required syntax for a TLS/SSL url is: +
tls://hostname:port[?options] +
The following parameters can be set via command line options
+(or in code via AVOptions):
+
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. +
+If enabled, try to verify the peer that we are communicating with. +Note, if using OpenSSL, this currently only makes sure that the +peer certificate is signed by one of the root certificates in the CA +database, but it does not validate that the certificate actually +matches the host name we are trying to connect to. (With other backends, +the host name is validated as well.) +
+This is disabled by default since it requires a CA database to be +provided by the caller in many cases. +
+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.) +
+A file containing the private key for the certificate. +
+If enabled, listen for connections on the provided port, and assume +the server role in the handshake instead of the client role. +
+The HTTP proxy to tunnel through, e.g. http://example.com:1234.
+The proxy must support the CONNECT method.
+
Example command lines: +
+To create a TLS/SSL server that serves an input stream. +
+ffmpeg -i input -f format tls://hostname:port?listen&cert=server.crt&key=server.key +
To play back a stream from the TLS/SSL server using ffplay:
+
ffplay tls://hostname:port +
User Datagram Protocol. +
+The required syntax for an UDP URL is: +
udp://hostname:port[?options] +
options contains a list of &-separated options of the form key=val. +
+In case threading is enabled on the system, a circular buffer is used +to store the incoming data, which allows one to reduce loss of data due to +UDP socket buffer overruns. The fifo_size and +overrun_nonfatal options are related to this buffer. +
+The list of supported options follows. +
+Set the UDP maximum socket buffer size in bytes. This is used to set either +the receive or send buffer size, depending on what the socket is used for. +Default is 32 KB for output, 384 KB for input. See also fifo_size. +
+If set to nonzero, the output will have the specified constant bitrate if the +input has enough packets to sustain it. +
+When using bitrate this specifies the maximum number of bits in +packet bursts. +
+Override the local UDP port to bind with. +
+Local IP address of a network interface used for sending packets or joining +multicast groups. +
+Set the size in bytes of UDP packets. +
+Explicitly allow or disallow reusing UDP sockets. +
+Set the time to live value (for multicast only). +
+Initialize the UDP socket with 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.
+
Only receive packets sent from the specified addresses. In case of multicast, +also subscribe to multicast traffic coming from these addresses only. +
+Ignore packets sent from the specified addresses. In case of multicast, also +exclude the source addresses in the multicast subscription. +
+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. +
+Survive in case of UDP receiving circular buffer overrun. Default +value is 0. +
+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. +
+Explicitly allow or disallow UDP broadcasting. +
+Note that broadcasting may not work properly on networks having +a broadcast storm protection. +
ffmpeg to stream over UDP to a remote endpoint:
+ffmpeg -i input -f format udp://hostname:port +
ffmpeg to stream in mpegts format over UDP using 188
+sized UDP packets, using a large input buffer:
+ffmpeg -i input -f mpegts udp://hostname:port?pkt_size=188&buffer_size=65535 +
ffmpeg to receive over UDP from a remote endpoint:
+ffmpeg -i udp://[multicast-address]:port ... +
Unix local socket +
+The required syntax for a Unix socket URL is: +
+unix://filepath +
The following parameters can be set via command line options
+(or in code via AVOptions):
+
Timeout in ms. +
Create the Unix socket in listening mode. +
ZeroMQ asynchronous messaging using the libzmq library. +
+This library supports unicast streaming to multiple clients without relying on +an external server. +
+The required syntax for streaming or connecting to a stream is: +
zmq:tcp://ip-address:port +
Example: +Create a localhost stream on port 5555: +
ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555 +
Multiple clients may connect to the stream using: +
ffplay zmq:tcp://127.0.0.1:5555 +
Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern. +The server side binds to a port and publishes data. Clients connect to the +server (via IP address/port) and subscribe to the stream. The order in which +the server and client start generally does not matter. +
+ffmpeg must be compiled with the –enable-libzmq option to support +this protocol. +
+Options can be set on the ffmpeg/ffplay command
+line. The following options are supported:
+
Forces the maximum packet size for sending/receiving data. The default value is +131,072 bytes. On the server side, this sets the maximum size of sent packets +via ZeroMQ. On the clients, it sets an internal buffer size for receiving +packets. Note that pkt_size on the clients should be equal to or greater than +pkt_size on the server. Otherwise the received message may be truncated causing +decoding errors. +
+ffmpeg, ffplay, ffprobe, +libavformat +
+ + +The FFmpeg developers. +
+For details about the authorship, see the Git history of the project
+(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
+git log in the FFmpeg source directory, or browsing the
+online repository at http://source.ffmpeg.org.
+
Maintainers for the specific components are listed in the file +MAINTAINERS in the source code tree. +
+ ++ This document was generated using makeinfo. +
+