| FFMPEG-PROTOCOLS(1) FFMPEG-PROTOCOLS(1) | |
| NAME | |
| ffmpeg-protocols - FFmpeg protocols | |
| DESCRIPTION | |
| This document describes the input and output protocols provided by the | |
| libavformat library. | |
| PROTOCOL OPTIONS | |
| 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: | |
| protocol_whitelist list (input) | |
| 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 | |
| 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: | |
| rw_timeout | |
| Maximum time to wait for (network) read/write operations to | |
| complete, in microseconds. | |
| A description of the currently available protocols follows. | |
| amqp | |
| 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: | |
| exchange | |
| 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). | |
| routing_key | |
| 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. | |
| pkt_size | |
| 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. | |
| connection_timeout | |
| 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. | |
| delivery_mode mode | |
| Sets the delivery mode of each message sent to broker. The | |
| following values are accepted: | |
| persistent | |
| Delivery mode set to "persistent" (2). This is the default | |
| value. Messages may be written to the broker's disk depending | |
| on its setup. | |
| non-persistent | |
| Delivery mode set to "non-persistent" (1). Messages will stay | |
| in broker's memory unless the broker is under memory pressure. | |
| async | |
| 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 | |
| bluray | |
| Read BluRay playlist. | |
| The accepted options are: | |
| angle | |
| BluRay angle | |
| chapter | |
| Start chapter (1...N) | |
| playlist | |
| 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 | |
| cache | |
| Caching wrapper for input stream. | |
| Cache the input stream to temporary file. It brings seeking capability | |
| to live streams. | |
| The accepted options are: | |
| read_ahead_limit | |
| 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> | |
| concat | |
| 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. | |
| concatf | |
| 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 | |
| crypto | |
| AES-encrypted stream reading protocol. | |
| The accepted options are: | |
| key Set the AES decryption key binary block from given hexadecimal | |
| representation. | |
| iv Set the AES decryption initialization vector binary block from | |
| given hexadecimal representation. | |
| Accepted URL formats: | |
| crypto:<URL> | |
| crypto+<URL> | |
| data | |
| 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 | |
| fd | |
| 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: | |
| blocksize | |
| 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. | |
| fd Set file descriptor. | |
| file | |
| 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 | |
| Truncate existing files on write, if set to 1. A value of 0 | |
| prevents truncating. Default value is 1. | |
| blocksize | |
| 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. | |
| follow | |
| 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). | |
| seekable | |
| 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 | |
| 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. | |
| timeout | |
| 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. | |
| ftp-user | |
| Set a user to be used for authenticating to the FTP server. This is | |
| overridden by the user in the FTP URL. | |
| ftp-password | |
| 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. | |
| ftp-anonymous-password | |
| Password used when login as anonymous user. Typically an e-mail | |
| address should be used. | |
| ftp-write-seekable | |
| Control seekability of connection during encoding. If set to 1 the | |
| resource is supposed to be seekable, if set to 0 it is assumed not | |
| to be seekable. Default value is 0. | |
| 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 | |
| Gopher protocol. | |
| gophers | |
| Gophers protocol. | |
| The Gopher protocol with TLS encapsulation. | |
| hls | |
| Read Apple HTTP Live Streaming compliant segmented stream as a uniform | |
| one. The M3U8 playlists describing the segments can be remote HTTP | |
| resources or local files, accessed using the standard file protocol. | |
| The nested protocol is declared by specifying "+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 | |
| HTTP (Hyper Text Transfer Protocol). | |
| This protocol accepts the following options: | |
| seekable | |
| Control seekability of connection. If set to 1 the resource is | |
| supposed to be seekable, if set to 0 it is assumed not to be | |
| seekable, if set to -1 it will try to autodetect if it is seekable. | |
| Default value is -1. | |
| chunked_post | |
| If set to 1 use chunked Transfer-Encoding for posts, default is 1. | |
| content_type | |
| Set a specific content type for the POST messages or for listen | |
| mode. | |
| http_proxy | |
| set HTTP proxy to tunnel through e.g. http://example.com:1234 | |
| headers | |
| Set custom HTTP headers, can override built in default headers. The | |
| value must be a string encoding the headers. | |
| multiple_requests | |
| Use persistent connections if set to 1, default is 0. | |
| post_data | |
| Set custom HTTP post data. | |
| referer | |
| Set the Referer header. Include 'Referer: URL' header in HTTP | |
| request. | |
| user_agent | |
| Override the User-Agent header. If not specified the protocol will | |
| use a string describing the libavformat build. ("Lavf/<version>") | |
| reconnect_at_eof | |
| If set then eof is treated like an error and causes reconnection, | |
| this is useful for live / endless streams. | |
| reconnect_streamed | |
| If set then even streamed/non seekable streams will be reconnected | |
| on errors. | |
| reconnect_on_network_error | |
| Reconnect automatically in case of TCP/TLS errors during connect. | |
| reconnect_on_http_error | |
| 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'. | |
| reconnect_delay_max | |
| Sets the maximum delay in seconds after which to give up | |
| reconnecting | |
| mime_type | |
| Export the MIME type. | |
| http_version | |
| Exports the HTTP response version number. Usually "1.0" or "1.1". | |
| icy If set to 1 request ICY (SHOUTcast) metadata from the server. If | |
| the server supports this, the metadata has to be retrieved by the | |
| application by reading the icy_metadata_headers and | |
| icy_metadata_packet options. The default is 1. | |
| icy_metadata_headers | |
| If the server supports ICY metadata, this contains the ICY-specific | |
| HTTP reply headers, separated by newline characters. | |
| icy_metadata_packet | |
| 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. | |
| cookies | |
| Set the cookies to be sent in future requests. The format of each | |
| cookie is the same as the value of a Set-Cookie HTTP response | |
| field. Multiple cookies can be delimited by a newline character. | |
| offset | |
| Set initial byte offset. | |
| end_offset | |
| Try to limit the request to bytes preceding this offset. | |
| method | |
| 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. | |
| listen | |
| 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_expect_100 | |
| 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. | |
| auth_type | |
| 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. | |
| none | |
| Choose the HTTP authentication type automatically. This is the | |
| default. | |
| basic | |
| 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. | |
| HTTP Cookies | |
| 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 | |
| Icecast protocol (stream to Icecast servers) | |
| This protocol accepts the following options: | |
| ice_genre | |
| Set the stream genre. | |
| ice_name | |
| Set the stream name. | |
| ice_description | |
| Set the stream description. | |
| ice_url | |
| Set the stream website URL. | |
| ice_public | |
| Set if the stream should be public. The default is 0 (not public). | |
| user_agent | |
| Override the User-Agent header. If not specified a string of the | |
| form "Lavf/<version>" will be used. | |
| password | |
| Set the Icecast mountpoint password. | |
| content_type | |
| Set the stream content type. This must be set if it is different | |
| from audio/mpeg. | |
| legacy_icecast | |
| This enables support for Icecast versions < 2.4.0, that do not | |
| support the HTTP PUT method but the SOURCE method. | |
| tls Establish a TLS (HTTPS) connection to Icecast. | |
| icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint> | |
| ipfs | |
| 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: | |
| gateway | |
| 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> | |
| mmst | |
| MMS (Microsoft Media Server) protocol over TCP. | |
| mmsh | |
| MMS (Microsoft Media Server) protocol over HTTP. | |
| The required syntax is: | |
| mmsh://<server>[:<port>][/<app>][/<playpath>] | |
| md5 | |
| MD5 output protocol. | |
| Computes the MD5 hash of the data to be written, and on close writes | |
| this to the designated output or stdout if none is specified. It can be | |
| used to test muxers without writing an actual file. | |
| Some examples follow. | |
| # 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. | |
| pipe | |
| 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: | |
| blocksize | |
| 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. | |
| fd 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. | |
| prompeg | |
| 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: | |
| l=n The number of columns (4-20, LxD <= 100) | |
| d=n The number of rows (4-20, LxD <= 100) | |
| Example usage: | |
| -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port> | |
| rist | |
| Reliable Internet Streaming Transport protocol | |
| The accepted options are: | |
| rist_profile | |
| Supported values: | |
| simple | |
| main | |
| This one is default. | |
| advanced | |
| buffer_size | |
| 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. | |
| fifo_size | |
| 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). | |
| overrun_nonfatal=1|0 | |
| Survive in case of librist fifo buffer overrun. Default value is 0. | |
| pkt_size | |
| Set maximum packet size for sending data. 1316 by default. | |
| log_level | |
| 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. | |
| secret | |
| Set override of encryption secret, by default is unset. | |
| encryption | |
| Set encryption type, by default is disabled. Acceptable values are | |
| 128 and 256. | |
| rtmp | |
| Real-Time Messaging Protocol. | |
| The Real-Time Messaging Protocol (RTMP) is used for streaming | |
| multimedia content across a TCP/IP network. | |
| The required syntax is: | |
| rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>] | |
| The accepted parameters are: | |
| username | |
| An optional username (mostly for publishing). | |
| password | |
| An optional password (mostly for publishing). | |
| server | |
| The address of the RTMP server. | |
| port | |
| The number of the TCP port to use (by default is 1935). | |
| app It is the name of the application to access. It usually corresponds | |
| to the path where the application is installed on the RTMP server | |
| (e.g. /ondemand/, /flash/live/, etc.). You can override the value | |
| parsed from the URI through the "rtmp_app" option, too. | |
| playpath | |
| 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. | |
| listen | |
| Act as a server, listening for an incoming connection. | |
| timeout | |
| 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 "AVOption"s): | |
| rtmp_app | |
| Name of application to connect on the RTMP server. This option | |
| overrides the parameter specified in the URI. | |
| rtmp_buffer | |
| Set the client buffer time in milliseconds. The default is 3000. | |
| rtmp_conn | |
| 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. | |
| rtmp_enhanced_codecs | |
| 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. | |
| rtmp_flashver | |
| Version of the Flash plugin used to run the SWF player. The default | |
| is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 | |
| (compatible; <libavformat version>).) | |
| rtmp_flush_interval | |
| Number of packets flushed in the same request (RTMPT only). The | |
| default is 10. | |
| rtmp_live | |
| 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". | |
| rtmp_pageurl | |
| URL of the web page in which the media was embedded. By default no | |
| value will be sent. | |
| rtmp_playpath | |
| Stream identifier to play or to publish. This option overrides the | |
| parameter specified in the URI. | |
| rtmp_subscribe | |
| Name of live stream to subscribe to. By default no value will be | |
| sent. It is only sent if the option is specified or if rtmp_live | |
| is set to live. | |
| rtmp_swfhash | |
| SHA256 hash of the decompressed SWF file (32 bytes). | |
| rtmp_swfsize | |
| Size of the decompressed SWF file, required for SWFVerification. | |
| rtmp_swfurl | |
| URL of the SWF player for the media. By default no value will be | |
| sent. | |
| rtmp_swfverify | |
| URL to player swf file, compute hash/size automatically. | |
| rtmp_tcurl | |
| URL of the target stream. Defaults to proto://host[:port]/app. | |
| tcp_nodelay=1|0 | |
| 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/ | |
| rtmpe | |
| Encrypted Real-Time Messaging Protocol. | |
| The Encrypted Real-Time Messaging Protocol (RTMPE) is used for | |
| streaming multimedia content within standard cryptographic primitives, | |
| consisting of Diffie-Hellman key exchange and HMACSHA256, generating a | |
| pair of RC4 keys. | |
| rtmps | |
| Real-Time Messaging Protocol over a secure SSL connection. | |
| The Real-Time Messaging Protocol (RTMPS) is used for streaming | |
| multimedia content across an encrypted connection. | |
| rtmpt | |
| Real-Time Messaging Protocol tunneled through HTTP. | |
| The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used | |
| for streaming multimedia content within HTTP requests to traverse | |
| firewalls. | |
| rtmpte | |
| Encrypted Real-Time Messaging Protocol tunneled through HTTP. | |
| The Encrypted Real-Time Messaging Protocol tunneled through HTTP | |
| (RTMPTE) is used for streaming multimedia content within HTTP requests | |
| to traverse firewalls. | |
| rtmpts | |
| Real-Time Messaging Protocol tunneled through HTTPS. | |
| The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is | |
| used for streaming multimedia content within HTTPS requests to traverse | |
| firewalls. | |
| libsmbclient | |
| 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. | |
| timeout | |
| Set timeout in milliseconds of socket I/O operations used by the | |
| underlying low level operation. By default it is set to -1, which | |
| means that the timeout is not specified. | |
| truncate | |
| Truncate existing files on write, if set to 1. A value of 0 | |
| prevents truncating. Default value is 1. | |
| workgroup | |
| Set the workgroup used for making connections. By default workgroup | |
| is not specified. | |
| For more information see: <http://www.samba.org/>. | |
| libssh | |
| 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. | |
| timeout | |
| Set timeout of socket I/O operations used by the underlying low | |
| level operation. By default it is set to -1, which means that the | |
| timeout is not specified. | |
| truncate | |
| Truncate existing files on write, if set to 1. A value of 0 | |
| prevents truncating. Default value is 1. | |
| private_key | |
| 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 | |
| librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte | |
| Real-Time Messaging Protocol and its variants supported through | |
| librtmp. | |
| Requires the presence of the librtmp headers and library during | |
| configuration. You need to explicitly configure the build with | |
| "--enable-librtmp". If enabled this will replace the native RTMP | |
| protocol. | |
| This protocol provides most client functions and a few server functions | |
| needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP | |
| (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these | |
| encrypted types (RTMPTE, RTMPTS). | |
| The required syntax is: | |
| <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" | |
| rtp | |
| 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: | |
| ttl=n | |
| Set the TTL (Time-To-Live) value (for multicast only). | |
| rtcpport=n | |
| Set the remote RTCP port to n. | |
| localrtpport=n | |
| Set the local RTP port to n. | |
| localrtcpport=n' | |
| Set the local RTCP port to n. | |
| pkt_size=n | |
| Set max packet size (in bytes) to n. | |
| buffer_size=size | |
| Set the maximum UDP socket buffer size in bytes. | |
| connect=0|1 | |
| Do a "connect()" on the UDP socket (if set to 1) or not (if set to | |
| 0). | |
| sources=ip[,ip] | |
| List allowed source IP addresses. | |
| block=ip[,ip] | |
| List disallowed (blocked) source IP addresses. | |
| write_to_source=0|1 | |
| Send packets to the source address of the latest received packet | |
| (if set to 1) or to a default remote address (if set to 0). | |
| localport=n | |
| Set the local RTP port to n. | |
| localaddr=addr | |
| Local IP address of a network interface used for sending packets or | |
| joining multicast groups. | |
| timeout=n | |
| Set timeout (in microseconds) of socket I/O operations to n. | |
| This is a deprecated option. Instead, localrtpport should be used. | |
| Important notes: | |
| 1. If rtcpport is not set the RTCP port will be set to the RTP port | |
| value plus 1. | |
| 2. If localrtpport (the local RTP port) is not set any available port | |
| will be used for the local RTP and RTCP ports. | |
| 3. If localrtcpport (the local RTCP port) is not set it will be set to | |
| the local RTP port value plus 1. | |
| rtsp | |
| Real-Time Streaming Protocol. | |
| RTSP is not technically a protocol handler in libavformat, it is a | |
| demuxer and muxer. The demuxer supports both normal RTSP (with data | |
| transferred over RTP; this is used by e.g. Apple and Microsoft) and | |
| Real-RTSP (with data transferred over RDT). | |
| The muxer can be used to send a stream using RTSP ANNOUNCE to a server | |
| supporting it (currently Darwin Streaming Server and Mischa | |
| Spiegelmock's <https://github.com/revmischa/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 "AVOption"s or in "avformat_open_input". | |
| Muxer | |
| The following options are supported. | |
| rtsp_transport | |
| Set RTSP transport protocols. | |
| It accepts the following values: | |
| udp Use UDP as lower transport protocol. | |
| tcp Use TCP (interleaving within the RTSP control channel) as lower | |
| transport protocol. | |
| Default value is 0. | |
| rtsp_flags | |
| Set RTSP flags. | |
| The following values are accepted: | |
| latm | |
| Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC. | |
| rfc2190 | |
| Use RFC 2190 packetization instead of RFC 4629 for H.263. | |
| skip_rtcp | |
| Don't send RTCP sender reports. | |
| h264_mode0 | |
| Use mode 0 for H.264 in RTP. | |
| send_bye | |
| Send RTCP BYE packets when finishing. | |
| Default value is 0. | |
| min_port | |
| Set minimum local UDP port. Default value is 5000. | |
| max_port | |
| Set maximum local UDP port. Default value is 65000. | |
| buffer_size | |
| Set the maximum socket buffer size in bytes. | |
| pkt_size | |
| Set max send packet size (in bytes). Default value is 1472. | |
| Demuxer | |
| The following options are supported. | |
| initial_pause | |
| Do not start playing the stream immediately if set to 1. Default | |
| value is 0. | |
| rtsp_transport | |
| Set RTSP transport protocols. | |
| It accepts the following values: | |
| udp Use UDP as lower transport protocol. | |
| tcp Use TCP (interleaving within the RTSP control channel) as lower | |
| transport protocol. | |
| udp_multicast | |
| Use UDP multicast as lower transport protocol. | |
| http | |
| Use HTTP tunneling as lower transport protocol, which is useful | |
| for passing proxies. | |
| https | |
| 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. | |
| rtsp_flags | |
| Set RTSP flags. | |
| The following values are accepted: | |
| filter_src | |
| Accept packets only from negotiated peer address and port. | |
| listen | |
| Act as a server, listening for an incoming connection. | |
| prefer_tcp | |
| Try TCP for RTP transport first, if TCP is available as RTSP | |
| RTP transport. | |
| satip_raw | |
| 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. | |
| allowed_media_types | |
| Set media types to accept from the server. | |
| The following flags are accepted: | |
| video | |
| audio | |
| data | |
| subtitle | |
| By default it accepts all media types. | |
| min_port | |
| Set minimum local UDP port. Default value is 5000. | |
| max_port | |
| Set maximum local UDP port. Default value is 65000. | |
| listen_timeout | |
| 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. | |
| reorder_queue_size | |
| Set number of packets to buffer for handling of reordered packets. | |
| timeout | |
| Set socket TCP I/O timeout in microseconds. | |
| user_agent | |
| Override User-Agent header. If not specified, it defaults to the | |
| libavformat identifier string. | |
| buffer_size | |
| 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". | |
| Examples | |
| The following examples all make use of the ffplay and ffmpeg tools. | |
| o Watch a stream over UDP, with a max reordering delay of 0.5 | |
| seconds: | |
| ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 | |
| o Watch a stream tunneled over HTTP: | |
| ffplay -rtsp_transport http rtsp://server/video.mp4 | |
| o Send a stream in realtime to a RTSP server, for others to watch: | |
| ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp | |
| o Receive a stream in realtime: | |
| ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output> | |
| sap | |
| Session Announcement Protocol (RFC 2974). This is not technically a | |
| protocol handler in libavformat, it is a muxer and demuxer. It is used | |
| for signalling of RTP streams, by announcing the SDP for the streams | |
| regularly on a separate port. | |
| Muxer | |
| 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: | |
| announce_addr=address | |
| Specify the destination IP address for sending the announcements | |
| to. If omitted, the announcements are sent to the commonly used | |
| SAP announcement multicast address 224.2.127.254 (sap.mcast.net), | |
| or ff0e::2:7ffe if destination is an IPv6 address. | |
| announce_port=port | |
| Specify the port to send the announcements on, defaults to 9875 if | |
| not specified. | |
| ttl=ttl | |
| Specify the time to live value for the announcements and RTP | |
| packets, defaults to 255. | |
| same_port=0|1 | |
| If set to 1, send all RTP streams on the same port pair. If zero | |
| (the default), all streams are sent on unique ports, with each | |
| stream on a port 2 numbers higher than the previous. VLC/Live555 | |
| requires this to be set to 1, to be able to receive the stream. | |
| The RTP stack in libavformat for receiving requires all streams to | |
| be sent on unique ports. | |
| 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] | |
| Demuxer | |
| 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] | |
| sctp | |
| Stream Control Transmission Protocol. | |
| The accepted URL syntax is: | |
| sctp://<host>:<port>[?<options>] | |
| The protocol accepts the following options: | |
| listen | |
| If set to any value, listen for an incoming connection. Outgoing | |
| connection is done by default. | |
| max_streams | |
| Set the maximum number of streams. By default no limit is set. | |
| srt | |
| 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. | |
| connect_timeout=milliseconds | |
| 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). | |
| ffs=bytes | |
| 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. | |
| inputbw=bytes/seconds | |
| 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. | |
| iptos=tos | |
| IP Type of Service. Applies to sender only. Default value is 0xB8. | |
| ipttl=ttl | |
| IP Time To Live. Applies to sender only. Default value is 64. | |
| latency=microseconds | |
| 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. | |
| listen_timeout=microseconds | |
| Set socket listen timeout. | |
| maxbw=bytes/seconds | |
| 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) | |
| mode=caller|listener|rendezvous | |
| Connection mode. caller opens client connection. listener starts | |
| server to listen for incoming connections. rendezvous use Rendez- | |
| Vous connection mode. Default value is caller. | |
| mss=bytes | |
| 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. | |
| nakreport=1|0 | |
| If set to 1, Receiver will send `UMSG_LOSSREPORT` messages | |
| periodically until a lost packet is retransmitted or intentionally | |
| dropped. Default value is 1. | |
| oheadbw=percents | |
| Recovery bandwidth overhead above input rate, in percents. See | |
| inputbw. Default value is 25%. | |
| passphrase=string | |
| 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). | |
| enforced_encryption=1|0 | |
| 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. | |
| kmrefreshrate=packets | |
| 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". | |
| kmpreannounce=packets | |
| 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". | |
| snddropdelay=microseconds | |
| 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. | |
| payload_size=bytes | |
| 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. | |
| pkt_size=bytes | |
| Alias for payload_size. | |
| peerlatency=microseconds | |
| The latency value (as described in rcvlatency) that is set by the | |
| sender side as a minimum value for the receiver. | |
| pbkeylen=bytes | |
| 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. | |
| rcvlatency=microseconds | |
| 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. | |
| recv_buffer_size=bytes | |
| Set UDP receive buffer size, expressed in bytes. | |
| send_buffer_size=bytes | |
| Set UDP send buffer size, expressed in bytes. | |
| timeout=microseconds | |
| 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. | |
| tlpktdrop=1|0 | |
| 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. | |
| sndbuf=bytes | |
| Set send buffer size, expressed in bytes. | |
| rcvbuf=bytes | |
| Set receive buffer size, expressed in bytes. | |
| Receive buffer must not be greater than ffs. | |
| lossmaxttl=packets | |
| 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. | |
| minversion | |
| 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. | |
| streamid=string | |
| 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 doesnXt make sense in Rendezvous connection; the result | |
| might be that simply one side will override the value from the | |
| other side and itXs the matter of luck which one would win | |
| srt_streamid=string | |
| Alias for streamid to avoid conflict with ffmpeg command line | |
| option. | |
| smoother=live|file | |
| 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. | |
| messageapi=1|0 | |
| When set, this socket uses the Message API, otherwise it uses | |
| Buffer API. Note that in live mode (see transtype) thereXs 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. | |
| transtype=live|file | |
| 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 | |
| linger=seconds | |
| 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". | |
| tsbpd=1|0 | |
| 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>. | |
| srtp | |
| Secure Real-time Transport Protocol. | |
| The accepted options are: | |
| srtp_in_suite | |
| srtp_out_suite | |
| Select input and output encoding suites. | |
| Supported values: | |
| AES_CM_128_HMAC_SHA1_80 | |
| SRTP_AES128_CM_HMAC_SHA1_80 | |
| AES_CM_128_HMAC_SHA1_32 | |
| SRTP_AES128_CM_HMAC_SHA1_32 | |
| srtp_in_params | |
| srtp_out_params | |
| Set input and output encoding parameters, which are expressed by a | |
| base64-encoded representation of a binary block. The first 16 bytes | |
| of this binary block are used as master key, the following 14 bytes | |
| are used as master salt. | |
| subfile | |
| Virtually extract a segment of a file or another stream. The | |
| underlying stream must be seekable. | |
| Accepted options: | |
| start | |
| Start offset of the extracted segment, in bytes. | |
| end 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 | |
| tee | |
| 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 | |
| tcp | |
| 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=2|1|0 | |
| 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_addr=addr | |
| Local IP address of a network interface used for tcp socket | |
| connect. | |
| local_port=port | |
| Local port used for tcp socket connect. | |
| timeout=microseconds | |
| Set raise error timeout, expressed in microseconds. | |
| This option is only relevant in read mode: if no data arrived in | |
| more than this time interval, raise error. | |
| listen_timeout=milliseconds | |
| Set listen timeout, expressed in milliseconds. | |
| recv_buffer_size=bytes | |
| Set receive buffer size, expressed bytes. | |
| send_buffer_size=bytes | |
| Set send buffer size, expressed bytes. | |
| tcp_nodelay=1|0 | |
| 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. | |
| tcp_mss=bytes | |
| 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> | |
| tls | |
| 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 "AVOption"s): | |
| ca_file, cafile=filename | |
| A file containing certificate authority (CA) root certificates to | |
| treat as trusted. If the linked TLS library contains a default this | |
| might not need to be specified for verification to work, but not | |
| all libraries and setups have defaults built in. The file must be | |
| in OpenSSL PEM format. | |
| tls_verify=1|0 | |
| If enabled, try to verify the peer that we are communicating with. | |
| Note, if using OpenSSL, this currently only makes sure that the | |
| peer certificate is signed by one of the root certificates in the | |
| CA database, but it does not validate that the certificate actually | |
| matches the host name we are trying to connect to. (With 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. | |
| cert_file, cert=filename | |
| A file containing a certificate to use in the handshake with the | |
| peer. (When operating as server, in listen mode, this is more | |
| often required by the peer, while client certificates only are | |
| mandated in certain setups.) | |
| key_file, key=filename | |
| A file containing the private key for the certificate. | |
| listen=1|0 | |
| If enabled, listen for connections on the provided port, and assume | |
| the server role in the handshake instead of the client role. | |
| http_proxy | |
| 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> | |
| udp | |
| 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. | |
| buffer_size=size | |
| Set the UDP maximum socket buffer size in bytes. This is used to | |
| set either the receive or send buffer size, depending on what the | |
| socket is used for. Default is 32 KB for output, 384 KB for input. | |
| See also fifo_size. | |
| bitrate=bitrate | |
| If set to nonzero, the output will have the specified constant | |
| bitrate if the input has enough packets to sustain it. | |
| burst_bits=bits | |
| When using bitrate this specifies the maximum number of bits in | |
| packet bursts. | |
| localport=port | |
| Override the local UDP port to bind with. | |
| localaddr=addr | |
| Local IP address of a network interface used for sending packets or | |
| joining multicast groups. | |
| pkt_size=size | |
| Set the size in bytes of UDP packets. | |
| reuse=1|0 | |
| Explicitly allow or disallow reusing UDP sockets. | |
| ttl=ttl | |
| Set the time to live value (for multicast only). | |
| connect=1|0 | |
| 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. | |
| sources=address[,address] | |
| Only receive packets sent from the specified addresses. In case of | |
| multicast, also subscribe to multicast traffic coming from these | |
| addresses only. | |
| block=address[,address] | |
| Ignore packets sent from the specified addresses. In case of | |
| multicast, also exclude the source addresses in the multicast | |
| subscription. | |
| fifo_size=units | |
| Set the UDP receiving circular buffer size, expressed as a number | |
| of packets with size of 188 bytes. If not specified defaults to | |
| 7*4096. | |
| overrun_nonfatal=1|0 | |
| Survive in case of UDP receiving circular buffer overrun. Default | |
| value is 0. | |
| timeout=microseconds | |
| Set raise error timeout, expressed in microseconds. | |
| This option is only relevant in read mode: if no data arrived in | |
| more than this time interval, raise error. | |
| broadcast=1|0 | |
| Explicitly allow or disallow UDP broadcasting. | |
| Note that broadcasting may not work properly on networks having a | |
| broadcast storm protection. | |
| Examples | |
| o Use ffmpeg to stream over UDP to a remote endpoint: | |
| ffmpeg -i <input> -f <format> udp://<hostname>:<port> | |
| o Use 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 | |
| o Use ffmpeg to receive over UDP from a remote endpoint: | |
| ffmpeg -i udp://[<multicast-address>]:<port> ... | |
| unix | |
| 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 "AVOption"s): | |
| timeout | |
| Timeout in ms. | |
| listen | |
| Create the Unix socket in listening mode. | |
| zmq | |
| 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: | |
| pkt_size | |
| 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. | |
| SEE ALSO | |
| ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3) | |
| AUTHORS | |
| 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. | |
| FFMPEG-PROTOCOLS(1) | |