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 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 descriptor access protocol. -
-The accepted syntax is: -
fd: -fd file_descriptor -
If fd is not specified, by default the stdout file descriptor will be -used for writing, stdin for reading. Unlike the pipe protocol, fd protocol has -seek support if it corresponding to a regular file. fd protocol doesn’t support -pass file descriptor via URL for security. -
-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.
-
Set file descriptor. -
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. -
-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 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. -
-Set a specific content type for the POST messages or for listen mode. -
-Override the User-Agent header. If not specified the protocol will use a -string describing the libavformat build. ("Lavf/<version>") -
-Set the Referer header. Include ’Referer: URL’ header in HTTP request. -
-Use persistent connections if set to 1, default is 0. -
-Set custom HTTP post data. -
-Export the MIME type. -
-Exports the HTTP response version number. Usually "1.0" or "1.1". -
-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. -
-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 an exported dictionary containing Icecast metadata from the bitstream, if present. -Only useful with the C API. -
-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. -
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. -
-An exported dictionary containing the content location. Only useful with the C -API. -
-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. -
-Reconnect automatically when disconnected before EOF is hit. -
-If set then eof is treated like an error and causes reconnection, this is useful -for live / endless streams. -
-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’. -
-If set then even streamed/non seekable streams will be reconnected on errors. -
-Set the maximum delay in seconds after which to give up reconnecting. -
-Set the maximum number of times to retry a connection. Default unset. -
-Set the maximum total delay in seconds after which to give up reconnecting. -
-If enabled, and a Retry-After header is encountered, its requested reconnection -delay will be honored, rather than using exponential backoff. Useful for 429 and -503 errors. Default enabled. -
-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 -
The resource requested by a client, when the experimental HTTP server is in use. -
-The HTTP code returned to the client, when the experimental HTTP server is in use. -
-Set the threshold, in bytes, for when a readahead should be prefered over a seek and -new HTTP request. This is useful, for example, to make sure the same connection -is used for reading large video packets with small audio packets in between. -
-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. -
-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.
-
One can use this protocol in 2 ways. Using IPFS: -
ffplay ipfs://<hash> -
Or the IPNS protocol (IPNS is mutable IPFS): -
ffplay ipns://<hash> -
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] -
If fd isn’t specified, 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.
-
Set file descriptor. -
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.
-
Specify the list of codecs the client advertises to support in an
-enhanced RTMP stream. This option should be set to a comma separated
-list of fourcc values, like hvc1,av01,vp09 for multiple codecs
-or hvc1 for only one codec. The specified list will be presented
-in the "fourCcLive" property of the Connect Command Message.
-
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. -
-Local IP address of a network interface used for tcp socket connect. -
-Local port used for tcp socket connect. -
-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
-(https://git.ffmpeg.org/ffmpeg), e.g. by typing the command
-git log in the FFmpeg source directory, or browsing the
-online repository at https://git.ffmpeg.org/ffmpeg.
-
Maintainers for the specific components are listed in the file -MAINTAINERS in the source code tree. -
- -- This document was generated using makeinfo. -
-