particular protocol using the option
"--disable-protocol=@var{PROTOCOL}".
-The option "-protocols" of the ff* tools will display the list of
+The option "-protocols" of the av* tools will display the list of
supported protocols.
-A description of the currently available protocols follows.
-
-@section hls
+All protocols accept the following options:
-Read Apple HTTP Live Streaming compliant segmented stream as
-a uniform one. The M3U8 playlists describing the segments can be
-remote HTTP resources or local files, accessed using the standard
-file protocol.
-The nested protocol is declared by specifying
-"+@var{proto}" after the hls URI scheme name, where @var{proto}
-is either "file" or "http".
+@table @option
+@item rw_timeout
+Maximum time to wait for (network) read/write operations to complete,
+in microseconds.
+@end table
-@example
-hls+http://host/path/to/remote/resource.m3u8
-hls+file://path/to/local/resource.m3u8
-@end example
+A description of the currently available protocols follows.
@section concat
protocol.
For example to read a sequence of files @file{split1.mpeg},
-@file{split2.mpeg}, @file{split3.mpeg} with @file{avplay} use the
+@file{split2.mpeg}, @file{split3.mpeg} with @command{avplay} use the
command:
@example
avplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
avconv -i file:input.mpeg output.mpeg
@end example
-The ff* tools default to the file protocol, that is a resource
+The av* tools default to the file protocol, that is a resource
specified with the name "FILE.mpeg" is interpreted as the URL
"file:FILE.mpeg".
+This protocol accepts the following options:
+
+@table @option
+@item 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).
+
+@end table
+
@section gopher
Gopher protocol.
+@section hls
+
+Read Apple HTTP Live Streaming compliant segmented stream as
+a uniform one. The M3U8 playlists describing the segments can be
+remote HTTP resources or local files, accessed using the standard
+file protocol.
+The nested protocol is declared by specifying
+"+@var{proto}" after the hls URI scheme name, where @var{proto}
+is either "file" or "http".
+
+@example
+hls+http://host/path/to/remote/resource.m3u8
+hls+file://path/to/local/resource.m3u8
+@end example
+
+Using this protocol is discouraged - the hls demuxer should work
+just as well (if not, please report the issues) and is more complete.
+To use the hls demuxer instead, simply use the direct URLs to the
+m3u8 files.
+
@section http
HTTP (Hyper Text Transfer Protocol).
+This protocol accepts the following options:
+
+@table @option
+@item chunked_post
+If set to 1 use chunked Transfer-Encoding for posts, default is 1.
+
+@item content_type
+Set a specific content type for the POST messages.
+
+@item headers
+Set custom HTTP headers, can override built in default headers. The
+value must be a string encoding the headers.
+
+@item multiple_requests
+Use persistent connections if set to 1, default is 0.
+
+@item post_data
+Set custom HTTP post data.
+
+@item user_agent
+Override the User-Agent header. If not specified a string of the form
+"Lavf/<version>" will be used.
+
+@item mime_type
+Export the MIME type.
+
+@item icy
+If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
+supports this, the metadata has to be retrieved by the application by reading
+the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
+The default is 1.
+
+@item icy_metadata_headers
+If the server supports ICY metadata, this contains the ICY-specific HTTP reply
+headers, separated by newline characters.
+
+@item icy_metadata_packet
+If the server supports ICY metadata, and @option{icy} was set to 1, this
+contains the last non-empty metadata packet sent by the server. It should be
+polled in regular intervals by applications interested in mid-stream metadata
+updates.
+
+@item offset
+Set initial byte offset.
+
+@item end_offset
+Try to limit the request to bytes preceding this offset.
+@end table
+
+@section Icecast
+
+Icecast (stream to Icecast servers)
+
+This protocol accepts the following options:
+
+@table @option
+@item ice_genre
+Set the stream genre.
+
+@item ice_name
+Set the stream name.
+
+@item ice_description
+Set the stream description.
+
+@item ice_url
+Set the stream website URL.
+
+@item ice_public
+Set if the stream should be public or not.
+The default is 0 (not public).
+
+@item user_agent
+Override the User-Agent header. If not specified a string of the form
+"Lavf/<version>" will be used.
+
+@item password
+Set the Icecast mountpoint password.
+
+@item content_type
+Set the stream content type. This must be set if it is different from
+audio/mpeg.
+
+@item legacy_icecast
+This enables support for Icecast versions < 2.4.0, that do not support the
+HTTP PUT method but the SOURCE method.
+
+@end table
+
@section mmst
MMS (Microsoft Media Server) protocol over TCP.
The required syntax is:
@example
-rtmp://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
+rtmp://[@var{username}:@var{password}@@]@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
@end example
The accepted parameters are:
@table @option
+@item username
+An optional username (mostly for publishing).
+
+@item password
+An optional password (mostly for publishing).
+
@item server
The address of the RTMP server.
@item app
It is the name of the application to access. It usually corresponds to
the path where the application is installed on the RTMP server
-(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.).
+(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
+the value parsed from the URI through the @code{rtmp_app} option, too.
@item playpath
It is the path or name of the resource to play with reference to the
-application specified in @var{app}, may be prefixed by "mp4:".
+application specified in @var{app}, may be prefixed by "mp4:". You
+can override the value parsed from the URI through the @code{rtmp_playpath}
+option, too.
+@item listen
+Act as a server, listening for an incoming connection.
+
+@item timeout
+Maximum time to wait for the incoming connection. Implies listen.
@end table
-For example to read with @file{avplay} a multimedia resource named
+Additionally, the following parameters can be set via command line options
+(or in code via @code{AVOption}s):
+@table @option
+
+@item rtmp_app
+Name of application to connect on the RTMP server. This option
+overrides the parameter specified in the URI.
+
+@item rtmp_buffer
+Set the client buffer time in milliseconds. The default is 3000.
+
+@item rtmp_conn
+Extra arbitrary AMF connection parameters, parsed from a string,
+e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
+Each value is prefixed by a single character denoting the type,
+B for Boolean, N for number, S for string, O for object, or Z for null,
+followed by a colon. For Booleans the data must be either 0 or 1 for
+FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
+1 to end or begin an object, respectively. Data items in subobjects may
+be named, by prefixing the type with 'N' and specifying the name before
+the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
+times to construct arbitrary AMF sequences.
+
+@item rtmp_flashver
+Version of the Flash plugin used to run the SWF player. The default
+is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible;
+<libavformat version>).)
+
+@item rtmp_flush_interval
+Number of packets flushed in the same request (RTMPT only). The default
+is 10.
+
+@item rtmp_live
+Specify that the media is a live stream. No resuming or seeking in
+live streams is possible. The default value is @code{any}, which means the
+subscriber first tries to play the live stream specified in the
+playpath. If a live stream of that name is not found, it plays the
+recorded stream. The other possible values are @code{live} and
+@code{recorded}.
+
+@item rtmp_pageurl
+URL of the web page in which the media was embedded. By default no
+value will be sent.
+
+@item rtmp_playpath
+Stream identifier to play or to publish. This option overrides the
+parameter specified in the URI.
+
+@item rtmp_subscribe
+Name of live stream to subscribe to. By default no value will be sent.
+It is only sent if the option is specified or if rtmp_live
+is set to live.
+
+@item rtmp_swfhash
+SHA256 hash of the decompressed SWF file (32 bytes).
+
+@item rtmp_swfsize
+Size of the decompressed SWF file, required for SWFVerification.
+
+@item rtmp_swfurl
+URL of the SWF player for the media. By default no value will be sent.
+
+@item rtmp_swfverify
+URL to player swf file, compute hash/size automatically.
+
+@item rtmp_tcurl
+URL of the target stream. Defaults to proto://host[:port]/app.
+
+@end table
+
+For example to read with @command{avplay} a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
@example
avplay rtmp://myserver/vod/sample
@end example
-@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
+To publish to a password protected server, passing the playpath and
+app names separately:
+@example
+avconv -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/
+@end example
+
+@section rtmpe
+
+Encrypted Real-Time Messaging Protocol.
+
+The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
+streaming multimedia content within standard cryptographic primitives,
+consisting of Diffie-Hellman key exchange and HMACSHA256, generating
+a pair of RC4 keys.
+
+@section rtmps
+
+Real-Time Messaging Protocol over a secure SSL connection.
+
+The Real-Time Messaging Protocol (RTMPS) is used for streaming
+multimedia content across an encrypted connection.
+
+@section rtmpt
+
+Real-Time Messaging Protocol tunneled through HTTP.
+
+The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
+for streaming multimedia content within HTTP requests to traverse
+firewalls.
+
+@section rtmpte
+
+Encrypted Real-Time Messaging Protocol tunneled through HTTP.
+
+The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
+is used for streaming multimedia content within HTTP requests to traverse
+firewalls.
+
+@section rtmpts
+
+Real-Time Messaging Protocol tunneled through HTTPS.
+
+The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
+for streaming multimedia content within HTTPS requests to traverse
+firewalls.
+
+@section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through
librtmp.
avconv -re -i myfile -f flv rtmp://myserver/live/mystream
@end example
-To play the same stream using @file{avplay}:
+To play the same stream using @command{avplay}:
@example
avplay "rtmp://myserver/live/mystream live=1"
@end example
rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example
-The following options (set on the @command{avconv}/@file{avplay} command
+The following options (set on the @command{avconv}/@command{avplay} command
line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
are supported:
@table @option
@item filter_src
Accept packets only from negotiated peer address and port.
+@item listen
+Act as a server, listening for an incoming connection.
@end table
When receiving data over UDP, the demuxer tries to reorder received packets
-(since they may arrive out of order, or packets may get lost totally). In
-order for this to be enabled, a maximum delay must be specified in the
-@code{max_delay} field of AVFormatContext.
+(since they may arrive out of order, or packets may get lost totally). This
+can be disabled by setting the maximum demuxing delay to zero (via
+the @code{max_delay} field of AVFormatContext).
-When watching multi-bitrate Real-RTSP streams with @file{avplay}, the
+When watching multi-bitrate Real-RTSP streams with @command{avplay}, the
streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
on the fly by pressing @code{v} and @code{a}.
avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example
+To receive a stream in realtime:
+
+@example
+avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
+@end example
+
@section sap
Session Announcement Protocol (RFC 2974). This is not technically a
avplay sap://[ff0e::2:7ffe]
@end example
+@section srt
+
+Haivision Secure Reliable Transport Protocol via libsrt.
+
+The supported syntax for a SRT URL is:
+@example
+srt://@var{hostname}:@var{port}[?@var{options}]
+@end example
+
+@var{options} contains a list of &-separated options of the form
+@var{key}=@var{val}.
+
+or
+
+@example
+@var{options} srt://@var{hostname}:@var{port}
+@end example
+
+@var{options} contains a list of '-@var{key} @var{val}'
+options.
+
+This protocol accepts the following options.
+
+@table @option
+@item connect_timeout
+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).
+
+@item ffs=@var{bytes}
+Flight Flag Size (Window Size), in bytes. FFS is actually an
+internal parameter and you should set it to not less than
+@option{recv_buffer_size} and @option{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.
+
+@item inputbw=@var{bytes/seconds}
+Sender nominal input rate, in bytes per seconds. Used along with
+@option{oheadbw}, when @option{maxbw} is set to relative (0), to
+calculate maximum sending rate when recovery packets are sent
+along with the main media stream:
+@option{inputbw} * (100 + @option{oheadbw}) / 100
+if @option{inputbw} is not set while @option{maxbw} is set to
+relative (0), the actual input rate is evaluated inside
+the library. Default value is 0.
+
+@item iptos=@var{tos}
+IP Type of Service. Applies to sender only. Default value is 0xB8.
+
+@item ipttl=@var{ttl}
+IP Time To Live. Applies to sender only. Default value is 64.
+
+@item listen_timeout
+Set socket listen timeout.
+
+@item maxbw=@var{bytes/seconds}
+Maximum sending bandwidth, in bytes per seconds.
+-1 infinite (CSRTCC limit is 30mbps)
+0 relative to input rate (see @option{inputbw})
+>0 absolute limit value
+Default value is 0 (relative)
+
+@item mode=@var{caller|listener|rendezvous}
+Connection mode.
+@option{caller} opens client connection.
+@option{listener} starts server to listen for incoming connections.
+@option{rendezvous} use Rendez-Vous connection mode.
+Default value is caller.
+
+@item mss=@var{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.
+
+@item nakreport=@var{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.
+
+@item oheadbw=@var{percents}
+Recovery bandwidth overhead above input rate, in percents.
+See @option{inputbw}. Default value is 25%.
+
+@item passphrase=@var{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 @option{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).
+
+@item pbkeylen=@var{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.
+
+@item recv_buffer_size=@var{bytes}
+Set receive buffer size, expressed in bytes.
+
+@item send_buffer_size=@var{bytes}
+Set send buffer size, expressed in bytes.
+
+@item rw_timeout
+Set raise error timeout for read/write optations.
+
+This option is only relevant in read mode:
+if no data arrived in more than this time
+interval, raise error.
+
+@item tlpktdrop=@var{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.
+
+@item tsbpddelay
+Timestamp-based Packet Delivery Delay.
+Used to absorb burst of missed packet retransmission.
+
+@end table
+
+For more information see: @url{https://github.com/Haivision/srt}.
+
@section tcp
-Trasmission Control Protocol.
+Transmission Control Protocol.
The required syntax for a TCP url is:
@example
@end table
+@section tls
+
+Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
+
+The required syntax for a TLS url is:
+@example
+tls://@var{hostname}:@var{port}
+@end example
+
+The following parameters can be set via command line options
+(or in code via @code{AVOption}s):
+
+@table @option
+
+@item ca_file
+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.
+
+@item tls_verify=@var{1|0}
+If enabled, try to verify the peer that we are communicating with.
+Note, if using OpenSSL, this currently only makes sure that the
+peer certificate is signed by one of the root certificates in the CA
+database, but it does not validate that the certificate actually
+matches the host name we are trying to connect to. (With GnuTLS,
+the host name is validated as well.)
+
+This is disabled by default since it requires a CA database to be
+provided by the caller in many cases.
+
+@item cert_file
+A file containing a certificate to use in the handshake with the peer.
+(When operating as server, in listen mode, this is more often required
+by the peer, while client certificates only are mandated in certain
+setups.)
+
+@item key_file
+A file containing the private key for the certificate.
+
+@item listen=@var{1|0}
+If enabled, listen for connections on the provided port, and assume
+the server role in the handshake instead of the client role.
+
+@end table
+
@section udp
User Datagram Protocol.
udp://@var{hostname}:@var{port}[?@var{options}]
@end example
-@var{options} contains a list of &-seperated options of the form @var{key}=@var{val}.
+@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
Follow the list of supported options.
@table @option
unreachable" is received.
For receiving, this gives the benefit of only receiving packets from
the specified peer address/port.
+
+@item sources=@var{address}[,@var{address}]
+Only receive packets sent to the multicast group from one of the
+specified sender IP addresses.
+
+@item block=@var{address}[,@var{address}]
+Ignore packets sent to the multicast group from the specified
+sender IP addresses.
@end table
Some usage examples of the udp protocol with @command{avconv} follow.
avconv -i udp://[@var{multicast-address}]:@var{port}
@end example
+@section unix
+
+Unix local socket
+
+The required syntax for a Unix socket URL is:
+
+@example
+unix://@var{filepath}
+@end example
+
+The following parameters can be set via command line options
+(or in code via @code{AVOption}s):
+
+@table @option
+@item timeout
+Timeout in ms.
+@item listen
+Create the Unix socket in listening mode.
+@end table
+
@c man end PROTOCOLS
projects / ffmpeg / blobdiff