forked from KolibriOS/kolibrios
1851 lines
36 KiB
Plaintext
1851 lines
36 KiB
Plaintext
=encoding utf8
|
|
|
|
=head1 NAME
|
|
|
|
ffmpeg-protocols - FFmpeg protocols
|
|
|
|
=head1 DESCRIPTION
|
|
|
|
|
|
This document describes the input and output protocols provided by the
|
|
libavformat library.
|
|
|
|
|
|
|
|
=head1 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=I<PROTOCOL>", or you can disable a
|
|
particular protocol using the option
|
|
"--disable-protocol=I<PROTOCOL>".
|
|
|
|
The option "-protocols" of the ff* tools will display the list of
|
|
supported protocols.
|
|
|
|
A description of the currently available protocols follows.
|
|
|
|
|
|
=head2 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
|
|
|
|
|
|
|
|
=head2 bluray
|
|
|
|
|
|
Read BluRay playlist.
|
|
|
|
The accepted options are:
|
|
|
|
=over 4
|
|
|
|
|
|
|
|
=item B<angle>
|
|
|
|
BluRay angle
|
|
|
|
|
|
=item B<chapter>
|
|
|
|
Start chapter (1...N)
|
|
|
|
|
|
=item B<playlist>
|
|
|
|
Playlist to read (BDMV/PLAYLIST/?????.mpls)
|
|
|
|
|
|
=back
|
|
|
|
|
|
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
|
|
|
|
|
|
|
|
=head2 cache
|
|
|
|
|
|
Caching wrapper for input stream.
|
|
|
|
Cache the input stream to temporary file. It brings seeking capability to live streams.
|
|
|
|
|
|
cache:<URL>
|
|
|
|
|
|
|
|
=head2 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 I<URL1>, I<URL2>, ..., I<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 F<split1.mpeg>,
|
|
F<split2.mpeg>, F<split3.mpeg> with B<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.
|
|
|
|
|
|
=head2 crypto
|
|
|
|
|
|
AES-encrypted stream reading protocol.
|
|
|
|
The accepted options are:
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<key>
|
|
|
|
Set the AES decryption key binary block from given hexadecimal representation.
|
|
|
|
|
|
=item B<iv>
|
|
|
|
Set the AES decryption initialization vector binary block from given hexadecimal representation.
|
|
|
|
=back
|
|
|
|
|
|
Accepted URL formats:
|
|
|
|
crypto:<URL>
|
|
crypto+<URL>
|
|
|
|
|
|
|
|
=head2 data
|
|
|
|
|
|
Data in-line in the URI. See E<lt>B<http://en.wikipedia.org/wiki/Data_URI_scheme>E<gt>.
|
|
|
|
For example, to convert a GIF file given inline with B<ffmpeg>:
|
|
|
|
ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
|
|
|
|
|
|
|
|
=head2 file
|
|
|
|
|
|
File access protocol.
|
|
|
|
Read from or write to a file.
|
|
|
|
A file URL can have the form:
|
|
|
|
file:<filename>
|
|
|
|
|
|
where I<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 F<input.mpeg> with B<ffmpeg>
|
|
use the command:
|
|
|
|
ffmpeg -i file:input.mpeg output.mpeg
|
|
|
|
|
|
This protocol accepts the following options:
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<truncate>
|
|
|
|
Truncate existing files on write, if set to 1. A value of 0 prevents
|
|
truncating. Default value is 1.
|
|
|
|
|
|
=item B<blocksize>
|
|
|
|
Set I/O operation maximum block size, in bytes. Default value is
|
|
C<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.
|
|
|
|
=back
|
|
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<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.
|
|
|
|
|
|
=item B<ftp-anonymous-password>
|
|
|
|
Password used when login as anonymous user. Typically an e-mail address
|
|
should be used.
|
|
|
|
|
|
=item B<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.
|
|
|
|
=back
|
|
|
|
|
|
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.
|
|
|
|
|
|
=head2 gopher
|
|
|
|
|
|
Gopher protocol.
|
|
|
|
|
|
=head2 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
|
|
"+I<proto>" after the hls URI scheme name, where I<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.
|
|
|
|
|
|
=head2 http
|
|
|
|
|
|
HTTP (Hyper Text Transfer Protocol).
|
|
|
|
This protocol accepts the following options:
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<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.
|
|
|
|
|
|
=item B<chunked_post>
|
|
|
|
If set to 1 use chunked Transfer-Encoding for posts, default is 1.
|
|
|
|
|
|
=item B<content_type>
|
|
|
|
Set a specific content type for the POST messages.
|
|
|
|
|
|
=item B<headers>
|
|
|
|
Set custom HTTP headers, can override built in default headers. The
|
|
value must be a string encoding the headers.
|
|
|
|
|
|
=item B<multiple_requests>
|
|
|
|
Use persistent connections if set to 1, default is 0.
|
|
|
|
|
|
=item B<post_data>
|
|
|
|
Set custom HTTP post data.
|
|
|
|
|
|
=item B<user-agent>
|
|
|
|
|
|
=item B<user_agent>
|
|
|
|
Override the User-Agent header. If not specified the protocol will use a
|
|
string describing the libavformat build. ("Lavf/E<lt>versionE<gt>")
|
|
|
|
|
|
=item B<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.
|
|
|
|
|
|
=item B<mime_type>
|
|
|
|
Export the MIME type.
|
|
|
|
|
|
=item B<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 B<icy_metadata_headers> and B<icy_metadata_packet> options.
|
|
The default is 1.
|
|
|
|
|
|
=item B<icy_metadata_headers>
|
|
|
|
If the server supports ICY metadata, this contains the ICY-specific HTTP reply
|
|
headers, separated by newline characters.
|
|
|
|
|
|
=item B<icy_metadata_packet>
|
|
|
|
If the server supports ICY metadata, and B<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 B<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.
|
|
|
|
|
|
=item B<offset>
|
|
|
|
Set initial byte offset.
|
|
|
|
|
|
=item B<end_offset>
|
|
|
|
Try to limit the request to bytes preceding this offset.
|
|
|
|
|
|
=item B<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.
|
|
|
|
|
|
=item B<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 mutli-client HTTP server. This is not yet implemented
|
|
in ffmpeg.c or ffserver.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>
|
|
|
|
|
|
|
|
=back
|
|
|
|
|
|
|
|
=head3 HTTP Cookies
|
|
|
|
|
|
Some HTTP requests will be denied unless cookie values are passed in with the
|
|
request. The B<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
|
|
|
|
|
|
|
|
=head2 Icecast
|
|
|
|
|
|
Icecast protocol (stream to Icecast servers)
|
|
|
|
This protocol accepts the following options:
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<ice_genre>
|
|
|
|
Set the stream genre.
|
|
|
|
|
|
=item B<ice_name>
|
|
|
|
Set the stream name.
|
|
|
|
|
|
=item B<ice_description>
|
|
|
|
Set the stream description.
|
|
|
|
|
|
=item B<ice_url>
|
|
|
|
Set the stream website URL.
|
|
|
|
|
|
=item B<ice_public>
|
|
|
|
Set if the stream should be public.
|
|
The default is 0 (not public).
|
|
|
|
|
|
=item B<user_agent>
|
|
|
|
Override the User-Agent header. If not specified a string of the form
|
|
"Lavf/E<lt>versionE<gt>" will be used.
|
|
|
|
|
|
=item B<password>
|
|
|
|
Set the Icecast mountpoint password.
|
|
|
|
|
|
=item B<content_type>
|
|
|
|
Set the stream content type. This must be set if it is different from
|
|
audio/mpeg.
|
|
|
|
|
|
=item B<legacy_icecast>
|
|
|
|
This enables support for Icecast versions E<lt> 2.4.0, that do not support the
|
|
HTTP PUT method but the SOURCE method.
|
|
|
|
|
|
=back
|
|
|
|
|
|
|
|
icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>
|
|
|
|
|
|
|
|
=head2 mmst
|
|
|
|
|
|
MMS (Microsoft Media Server) protocol over TCP.
|
|
|
|
|
|
=head2 mmsh
|
|
|
|
|
|
MMS (Microsoft Media Server) protocol over HTTP.
|
|
|
|
The required syntax is:
|
|
|
|
mmsh://<server>[:<port>][/<app>][/<playpath>]
|
|
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=head2 pipe
|
|
|
|
|
|
UNIX pipe access protocol.
|
|
|
|
Read and write from UNIX pipes.
|
|
|
|
The accepted syntax is:
|
|
|
|
pipe:[<number>]
|
|
|
|
|
|
I<number> is the number corresponding to the file descriptor of the
|
|
pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If I<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 B<ffmpeg>:
|
|
|
|
cat test.wav | ffmpeg -i pipe:0
|
|
# ...this is the same as...
|
|
cat test.wav | ffmpeg -i pipe:
|
|
|
|
|
|
For writing to stdout with B<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:
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<blocksize>
|
|
|
|
Set I/O operation maximum block size, in bytes. Default value is
|
|
C<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.
|
|
|
|
=back
|
|
|
|
|
|
Note that some formats (typically MOV), require the output protocol to
|
|
be seekable, so they will fail with the pipe output protocol.
|
|
|
|
|
|
=head2 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:
|
|
|
|
=over 4
|
|
|
|
|
|
|
|
=item B<username>
|
|
|
|
An optional username (mostly for publishing).
|
|
|
|
|
|
=item B<password>
|
|
|
|
An optional password (mostly for publishing).
|
|
|
|
|
|
=item B<server>
|
|
|
|
The address of the RTMP server.
|
|
|
|
|
|
=item B<port>
|
|
|
|
The number of the TCP port to use (by default is 1935).
|
|
|
|
|
|
=item B<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. F</ondemand/>, F</flash/live/>, etc.). You can override
|
|
the value parsed from the URI through the C<rtmp_app> option, too.
|
|
|
|
|
|
=item B<playpath>
|
|
|
|
It is the path or name of the resource to play with reference to the
|
|
application specified in I<app>, may be prefixed by "mp4:". You
|
|
can override the value parsed from the URI through the C<rtmp_playpath>
|
|
option, too.
|
|
|
|
|
|
=item B<listen>
|
|
|
|
Act as a server, listening for an incoming connection.
|
|
|
|
|
|
=item B<timeout>
|
|
|
|
Maximum time to wait for the incoming connection. Implies listen.
|
|
|
|
=back
|
|
|
|
|
|
Additionally, the following parameters can be set via command line options
|
|
(or in code via C<AVOption>s):
|
|
|
|
=over 4
|
|
|
|
|
|
|
|
=item B<rtmp_app>
|
|
|
|
Name of application to connect on the RTMP server. This option
|
|
overrides the parameter specified in the URI.
|
|
|
|
|
|
=item B<rtmp_buffer>
|
|
|
|
Set the client buffer time in milliseconds. The default is 3000.
|
|
|
|
|
|
=item B<rtmp_conn>
|
|
|
|
Extra arbitrary AMF connection parameters, parsed from a string,
|
|
e.g. like C<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. C<NB:myFlag:1>). This option may be used multiple
|
|
times to construct arbitrary AMF sequences.
|
|
|
|
|
|
=item B<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;
|
|
E<lt>libavformat versionE<gt>).)
|
|
|
|
|
|
=item B<rtmp_flush_interval>
|
|
|
|
Number of packets flushed in the same request (RTMPT only). The default
|
|
is 10.
|
|
|
|
|
|
=item B<rtmp_live>
|
|
|
|
Specify that the media is a live stream. No resuming or seeking in
|
|
live streams is possible. The default value is C<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 C<live> and
|
|
C<recorded>.
|
|
|
|
|
|
=item B<rtmp_pageurl>
|
|
|
|
URL of the web page in which the media was embedded. By default no
|
|
value will be sent.
|
|
|
|
|
|
=item B<rtmp_playpath>
|
|
|
|
Stream identifier to play or to publish. This option overrides the
|
|
parameter specified in the URI.
|
|
|
|
|
|
=item B<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 B<rtmp_swfhash>
|
|
|
|
SHA256 hash of the decompressed SWF file (32 bytes).
|
|
|
|
|
|
=item B<rtmp_swfsize>
|
|
|
|
Size of the decompressed SWF file, required for SWFVerification.
|
|
|
|
|
|
=item B<rtmp_swfurl>
|
|
|
|
URL of the SWF player for the media. By default no value will be sent.
|
|
|
|
|
|
=item B<rtmp_swfverify>
|
|
|
|
URL to player swf file, compute hash/size automatically.
|
|
|
|
|
|
=item B<rtmp_tcurl>
|
|
|
|
URL of the target stream. Defaults to proto://host[:port]/app.
|
|
|
|
|
|
=back
|
|
|
|
|
|
For example to read with B<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/
|
|
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<timeout>
|
|
|
|
Set timeout in miliseconds 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.
|
|
|
|
|
|
=item B<truncate>
|
|
|
|
Truncate existing files on write, if set to 1. A value of 0 prevents
|
|
truncating. Default value is 1.
|
|
|
|
|
|
=item B<workgroup>
|
|
|
|
Set the workgroup used for making connections. By default workgroup is not specified.
|
|
|
|
|
|
=back
|
|
|
|
|
|
For more information see: E<lt>B<http://www.samba.org/>E<gt>.
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<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.
|
|
|
|
|
|
=item B<truncate>
|
|
|
|
Truncate existing files on write, if set to 1. A value of 0 prevents
|
|
truncating. Default value is 1.
|
|
|
|
|
|
=item B<private_key>
|
|
|
|
Specify the path of the file containing private key to use during authorization.
|
|
By default libssh searches for keys in the F<~/.ssh/> directory.
|
|
|
|
|
|
=back
|
|
|
|
|
|
Example: Play a file stored on remote server.
|
|
|
|
|
|
ffplay sftp://user:password@server_address:22/home/user/resource.mpeg
|
|
|
|
|
|
|
|
=head2 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 I<rtmp_proto> is one of the strings "rtmp", "rtmpt", "rtmpe",
|
|
"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
|
|
I<server>, I<port>, I<app> and I<playpath> have the same
|
|
meaning as specified for the RTMP native protocol.
|
|
I<options> contains a list of space-separated options of the form
|
|
I<key>=I<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
|
|
B<ffmpeg>:
|
|
|
|
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
|
|
|
|
|
|
To play the same stream using B<ffplay>:
|
|
|
|
ffplay "rtmp://myserver/live/mystream live=1"
|
|
|
|
|
|
|
|
=head2 rtp
|
|
|
|
|
|
Real-time Transport Protocol.
|
|
|
|
The required syntax for an RTP URL is:
|
|
rtp://I<hostname>[:I<port>][?I<option>=I<val>...]
|
|
|
|
I<port> specifies the RTP port to use.
|
|
|
|
The following URL options are supported:
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
|
|
=item B<ttl=>I<n>
|
|
|
|
Set the TTL (Time-To-Live) value (for multicast only).
|
|
|
|
|
|
=item B<rtcpport=>I<n>
|
|
|
|
Set the remote RTCP port to I<n>.
|
|
|
|
|
|
=item B<localrtpport=>I<n>
|
|
|
|
Set the local RTP port to I<n>.
|
|
|
|
|
|
=item B<localrtcpport=>I<n>B<'>
|
|
|
|
Set the local RTCP port to I<n>.
|
|
|
|
|
|
=item B<pkt_size=>I<n>
|
|
|
|
Set max packet size (in bytes) to I<n>.
|
|
|
|
|
|
=item B<connect=0|1>
|
|
|
|
Do a C<connect()> on the UDP socket (if set to 1) or not (if set
|
|
to 0).
|
|
|
|
|
|
=item B<sources=>I<ip>B<[,>I<ip>B<]>
|
|
|
|
List allowed source IP addresses.
|
|
|
|
|
|
=item B<block=>I<ip>B<[,>I<ip>B<]>
|
|
|
|
List disallowed (blocked) source IP addresses.
|
|
|
|
|
|
=item B<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).
|
|
|
|
|
|
=item B<localport=>I<n>
|
|
|
|
Set the local RTP port to I<n>.
|
|
|
|
This is a deprecated option. Instead, B<localrtpport> should be
|
|
used.
|
|
|
|
|
|
=back
|
|
|
|
|
|
Important notes:
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
|
|
=item 1.
|
|
|
|
If B<rtcpport> is not set the RTCP port will be set to the RTP
|
|
port value plus 1.
|
|
|
|
|
|
=item 2.
|
|
|
|
If B<localrtpport> (the local RTP port) is not set any available
|
|
port will be used for the local RTP and RTCP ports.
|
|
|
|
|
|
=item 3.
|
|
|
|
If B<localrtcpport> (the local RTCP port) is not set it will be
|
|
set to the local RTP port value plus 1.
|
|
|
|
=back
|
|
|
|
|
|
|
|
=head2 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
|
|
E<lt>B<https://github.com/revmischa/rtsp-server>E<gt>).
|
|
|
|
The required syntax for a RTSP url is:
|
|
|
|
rtsp://<hostname>[:<port>]/<path>
|
|
|
|
|
|
Options can be set on the B<ffmpeg>/B<ffplay> command
|
|
line, or set in code via C<AVOption>s or in
|
|
C<avformat_open_input>.
|
|
|
|
The following options are supported.
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<initial_pause>
|
|
|
|
Do not start playing the stream immediately if set to 1. Default value
|
|
is 0.
|
|
|
|
|
|
=item B<rtsp_transport>
|
|
|
|
Set RTSP transport protocols.
|
|
|
|
It accepts the following values:
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<udp>
|
|
|
|
Use UDP as lower transport protocol.
|
|
|
|
|
|
=item B<tcp>
|
|
|
|
Use TCP (interleaving within the RTSP control channel) as lower
|
|
transport protocol.
|
|
|
|
|
|
=item B<udp_multicast>
|
|
|
|
Use UDP multicast as lower transport protocol.
|
|
|
|
|
|
=item B<http>
|
|
|
|
Use HTTP tunneling as lower transport protocol, which is useful for
|
|
passing proxies.
|
|
|
|
=back
|
|
|
|
|
|
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 B<tcp> and B<udp> options are supported.
|
|
|
|
|
|
=item B<rtsp_flags>
|
|
|
|
Set RTSP flags.
|
|
|
|
The following values are accepted:
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<filter_src>
|
|
|
|
Accept packets only from negotiated peer address and port.
|
|
|
|
=item B<listen>
|
|
|
|
Act as a server, listening for an incoming connection.
|
|
|
|
=item B<prefer_tcp>
|
|
|
|
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
|
|
|
|
=back
|
|
|
|
|
|
Default value is B<none>.
|
|
|
|
|
|
=item B<allowed_media_types>
|
|
|
|
Set media types to accept from the server.
|
|
|
|
The following flags are accepted:
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<video>
|
|
|
|
|
|
=item B<audio>
|
|
|
|
|
|
=item B<data>
|
|
|
|
|
|
=back
|
|
|
|
|
|
By default it accepts all media types.
|
|
|
|
|
|
=item B<min_port>
|
|
|
|
Set minimum local UDP port. Default value is 5000.
|
|
|
|
|
|
=item B<max_port>
|
|
|
|
Set maximum local UDP port. Default value is 65000.
|
|
|
|
|
|
=item B<timeout>
|
|
|
|
Set maximum timeout (in seconds) to wait for incoming connections.
|
|
|
|
A value of -1 means infinite (default). This option implies the
|
|
B<rtsp_flags> set to B<listen>.
|
|
|
|
|
|
=item B<reorder_queue_size>
|
|
|
|
Set number of packets to buffer for handling of reordered packets.
|
|
|
|
|
|
=item B<stimeout>
|
|
|
|
Set socket TCP I/O timeout in microseconds.
|
|
|
|
|
|
=item B<user-agent>
|
|
|
|
Override User-Agent header. If not specified, it defaults to the
|
|
libavformat identifier string.
|
|
|
|
=back
|
|
|
|
|
|
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 C<max_delay> field of AVFormatContext).
|
|
|
|
When watching multi-bitrate Real-RTSP streams with B<ffplay>, the
|
|
streams to display can be chosen with C<-vst> I<n> and
|
|
C<-ast> I<n> for video and audio respectively, and can be switched
|
|
on the fly by pressing C<v> and C<a>.
|
|
|
|
|
|
=head3 Examples
|
|
|
|
|
|
The following examples all make use of the B<ffplay> and
|
|
B<ffmpeg> tools.
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item *
|
|
|
|
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
|
|
|
|
|
|
|
|
=item *
|
|
|
|
Watch a stream tunneled over HTTP:
|
|
|
|
ffplay -rtsp_transport http rtsp://server/video.mp4
|
|
|
|
|
|
|
|
=item *
|
|
|
|
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
|
|
|
|
|
|
|
|
=item *
|
|
|
|
Receive a stream in realtime:
|
|
|
|
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>
|
|
|
|
|
|
=back
|
|
|
|
|
|
|
|
=head2 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.
|
|
|
|
|
|
=head3 Muxer
|
|
|
|
|
|
The syntax for a SAP url given to the muxer is:
|
|
|
|
sap://<destination>[:<port>][?<options>]
|
|
|
|
|
|
The RTP packets are sent to I<destination> on port I<port>,
|
|
or to port 5004 if no port is specified.
|
|
I<options> is a C<&>-separated list. The following options
|
|
are supported:
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
|
|
=item B<announce_addr=>I<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 I<destination> is an IPv6 address.
|
|
|
|
|
|
=item B<announce_port=>I<port>
|
|
|
|
Specify the port to send the announcements on, defaults to
|
|
9875 if not specified.
|
|
|
|
|
|
=item B<ttl=>I<ttl>
|
|
|
|
Specify the time to live value for the announcements and RTP packets,
|
|
defaults to 255.
|
|
|
|
|
|
=item B<same_port=>I<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.
|
|
|
|
=back
|
|
|
|
|
|
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 B<ffplay>:
|
|
|
|
|
|
ffmpeg -re -i <input> -f sap sap://224.0.0.255
|
|
|
|
|
|
And for watching in B<ffplay>, over IPv6:
|
|
|
|
|
|
ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]
|
|
|
|
|
|
|
|
=head3 Demuxer
|
|
|
|
|
|
The syntax for a SAP url given to the demuxer is:
|
|
|
|
sap://[<address>][:<port>]
|
|
|
|
|
|
I<address> is the multicast address to listen for announcements on,
|
|
if omitted, the default 224.2.127.254 (sap.mcast.net) is used. I<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]
|
|
|
|
|
|
|
|
=head2 sctp
|
|
|
|
|
|
Stream Control Transmission Protocol.
|
|
|
|
The accepted URL syntax is:
|
|
|
|
sctp://<host>:<port>[?<options>]
|
|
|
|
|
|
The protocol accepts the following options:
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<listen>
|
|
|
|
If set to any value, listen for an incoming connection. Outgoing connection is done by default.
|
|
|
|
|
|
=item B<max_streams>
|
|
|
|
Set the maximum number of streams. By default no limit is set.
|
|
|
|
=back
|
|
|
|
|
|
|
|
=head2 srtp
|
|
|
|
|
|
Secure Real-time Transport Protocol.
|
|
|
|
The accepted options are:
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<srtp_in_suite>
|
|
|
|
|
|
=item B<srtp_out_suite>
|
|
|
|
Select input and output encoding suites.
|
|
|
|
Supported values:
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<AES_CM_128_HMAC_SHA1_80>
|
|
|
|
|
|
=item B<SRTP_AES128_CM_HMAC_SHA1_80>
|
|
|
|
|
|
=item B<AES_CM_128_HMAC_SHA1_32>
|
|
|
|
|
|
=item B<SRTP_AES128_CM_HMAC_SHA1_32>
|
|
|
|
|
|
=back
|
|
|
|
|
|
|
|
=item B<srtp_in_params>
|
|
|
|
|
|
=item B<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.
|
|
|
|
=back
|
|
|
|
|
|
|
|
=head2 subfile
|
|
|
|
|
|
Virtually extract a segment of a file or another stream.
|
|
The underlying stream must be seekable.
|
|
|
|
Accepted options:
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<start>
|
|
|
|
Start offset of the extracted segment, in bytes.
|
|
|
|
=item B<end>
|
|
|
|
End offset of the extracted segment, in bytes.
|
|
|
|
=back
|
|
|
|
|
|
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
|
|
|
|
|
|
|
|
=head2 tcp
|
|
|
|
|
|
Transmission Control Protocol.
|
|
|
|
The required syntax for a TCP url is:
|
|
|
|
tcp://<hostname>:<port>[?<options>]
|
|
|
|
|
|
I<options> contains a list of &-separated options of the form
|
|
I<key>=I<val>.
|
|
|
|
The list of supported options follows.
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<listen=>I<1|0>
|
|
|
|
Listen for an incoming connection. Default value is 0.
|
|
|
|
|
|
=item B<timeout=>I<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.
|
|
|
|
|
|
=item B<listen_timeout=>I<milliseconds>
|
|
|
|
Set listen timeout, expressed in milliseconds.
|
|
|
|
=back
|
|
|
|
|
|
The following example shows how to setup a listening TCP connection
|
|
with B<ffmpeg>, which is then accessed with B<ffplay>:
|
|
|
|
ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
|
|
ffplay tcp://<hostname>:<port>
|
|
|
|
|
|
|
|
=head2 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 C<AVOption>s):
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
|
|
=item B<ca_file, cafile=>I<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.
|
|
|
|
|
|
=item B<tls_verify=>I<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 B<cert_file, cert=>I<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.)
|
|
|
|
|
|
=item B<key_file, key=>I<filename>
|
|
|
|
A file containing the private key for the certificate.
|
|
|
|
|
|
=item B<listen=>I<1|0>
|
|
|
|
If enabled, listen for connections on the provided port, and assume
|
|
the server role in the handshake instead of the client role.
|
|
|
|
|
|
=back
|
|
|
|
|
|
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 B<ffplay>:
|
|
|
|
|
|
ffplay tls://<hostname>:<port>
|
|
|
|
|
|
|
|
=head2 udp
|
|
|
|
|
|
User Datagram Protocol.
|
|
|
|
The required syntax for an UDP URL is:
|
|
|
|
udp://<hostname>:<port>[?<options>]
|
|
|
|
|
|
I<options> contains a list of &-separated options of the form I<key>=I<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 I<fifo_size> and
|
|
I<overrun_nonfatal> options are related to this buffer.
|
|
|
|
The list of supported options follows.
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<buffer_size=>I<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 64KB. See also I<fifo_size>.
|
|
|
|
|
|
=item B<localport=>I<port>
|
|
|
|
Override the local UDP port to bind with.
|
|
|
|
|
|
=item B<localaddr=>I<addr>
|
|
|
|
Choose the local IP address. This is useful e.g. if sending multicast
|
|
and the host has multiple interfaces, where the user can choose
|
|
which interface to send on by specifying the IP address of that interface.
|
|
|
|
|
|
=item B<pkt_size=>I<size>
|
|
|
|
Set the size in bytes of UDP packets.
|
|
|
|
|
|
=item B<reuse=>I<1|0>
|
|
|
|
Explicitly allow or disallow reusing UDP sockets.
|
|
|
|
|
|
=item B<ttl=>I<ttl>
|
|
|
|
Set the time to live value (for multicast only).
|
|
|
|
|
|
=item B<connect=>I<1|0>
|
|
|
|
Initialize the UDP socket with C<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.
|
|
|
|
|
|
=item B<sources=>I<address>B<[,>I<address>B<]>
|
|
|
|
Only receive packets sent to the multicast group from one of the
|
|
specified sender IP addresses.
|
|
|
|
|
|
=item B<block=>I<address>B<[,>I<address>B<]>
|
|
|
|
Ignore packets sent to the multicast group from the specified
|
|
sender IP addresses.
|
|
|
|
|
|
=item B<fifo_size=>I<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.
|
|
|
|
|
|
=item B<overrun_nonfatal=>I<1|0>
|
|
|
|
Survive in case of UDP receiving circular buffer overrun. Default
|
|
value is 0.
|
|
|
|
|
|
=item B<timeout=>I<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.
|
|
|
|
|
|
=item B<broadcast=>I<1|0>
|
|
|
|
Explicitly allow or disallow UDP broadcasting.
|
|
|
|
Note that broadcasting may not work properly on networks having
|
|
a broadcast storm protection.
|
|
|
|
=back
|
|
|
|
|
|
|
|
=head3 Examples
|
|
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item *
|
|
|
|
Use B<ffmpeg> to stream over UDP to a remote endpoint:
|
|
|
|
ffmpeg -i <input> -f <format> udp://<hostname>:<port>
|
|
|
|
|
|
|
|
=item *
|
|
|
|
Use B<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
|
|
|
|
|
|
|
|
=item *
|
|
|
|
Use B<ffmpeg> to receive over UDP from a remote endpoint:
|
|
|
|
ffmpeg -i udp://[<multicast-address>]:<port> ...
|
|
|
|
|
|
=back
|
|
|
|
|
|
|
|
=head2 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 C<AVOption>s):
|
|
|
|
|
|
=over 4
|
|
|
|
|
|
=item B<timeout>
|
|
|
|
Timeout in ms.
|
|
|
|
=item B<listen>
|
|
|
|
Create the Unix socket in listening mode.
|
|
|
|
=back
|
|
|
|
|
|
|
|
|
|
=head1 SEE ALSO
|
|
|
|
|
|
|
|
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
|
|
|
|
|
|
=head1 AUTHORS
|
|
|
|
|
|
The FFmpeg developers.
|
|
|
|
For details about the authorship, see the Git history of the project
|
|
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
|
|
B<git log> in the FFmpeg source directory, or browsing the
|
|
online repository at E<lt>B<http://source.ffmpeg.org>E<gt>.
|
|
|
|
Maintainers for the specific components are listed in the file
|
|
F<MAINTAINERS> in the source code tree.
|
|
|
|
|
|
|