mirror of https://github.com/FFmpeg/FFmpeg.git
You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
1456 lines
41 KiB
1456 lines
41 KiB
@chapter Protocol Options |
|
@c man begin 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 -@var{option} @var{value} in the |
|
FFmpeg tools, or by setting the value explicitly in the |
|
@code{AVFormatContext} options or using the @file{libavutil/opt.h} API |
|
for programmatic use. |
|
|
|
The list of supported options follows: |
|
|
|
@table @option |
|
@item protocol_whitelist @var{list} (@emph{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. |
|
@end table |
|
|
|
@c man end PROTOCOL OPTIONS |
|
|
|
@chapter Protocols |
|
@c man begin 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=@var{PROTOCOL}", or you can disable a |
|
particular protocol using the option |
|
"--disable-protocol=@var{PROTOCOL}". |
|
|
|
The option "-protocols" of the ff* tools will display the list of |
|
supported protocols. |
|
|
|
All protocols accept the following options: |
|
|
|
@table @option |
|
@item rw_timeout |
|
Maximum time to wait for (network) read/write operations to complete, |
|
in microseconds. |
|
@end table |
|
|
|
A description of the currently available protocols follows. |
|
|
|
@section async |
|
|
|
Asynchronous data filling wrapper for input stream. |
|
|
|
Fill data in a background thread, to decouple I/O operation from demux thread. |
|
|
|
@example |
|
async:@var{URL} |
|
async:http://host/resource |
|
async:cache:http://host/resource |
|
@end example |
|
|
|
@section bluray |
|
|
|
Read BluRay playlist. |
|
|
|
The accepted options are: |
|
@table @option |
|
|
|
@item angle |
|
BluRay angle |
|
|
|
@item chapter |
|
Start chapter (1...N) |
|
|
|
@item playlist |
|
Playlist to read (BDMV/PLAYLIST/?????.mpls) |
|
|
|
@end table |
|
|
|
Examples: |
|
|
|
Read longest playlist from BluRay mounted to /mnt/bluray: |
|
@example |
|
bluray:/mnt/bluray |
|
@end example |
|
|
|
Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: |
|
@example |
|
-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray |
|
@end example |
|
|
|
@section cache |
|
|
|
Caching wrapper for input stream. |
|
|
|
Cache the input stream to temporary file. It brings seeking capability to live streams. |
|
|
|
@example |
|
cache:@var{URL} |
|
@end example |
|
|
|
@section 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: |
|
@example |
|
concat:@var{URL1}|@var{URL2}|...|@var{URLN} |
|
@end example |
|
|
|
where @var{URL1}, @var{URL2}, ..., @var{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 @file{split1.mpeg}, |
|
@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the |
|
command: |
|
@example |
|
ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg |
|
@end example |
|
|
|
Note that you may need to escape the character "|" which is special for |
|
many shells. |
|
|
|
@section crypto |
|
|
|
AES-encrypted stream reading protocol. |
|
|
|
The accepted options are: |
|
@table @option |
|
@item key |
|
Set the AES decryption key binary block from given hexadecimal representation. |
|
|
|
@item iv |
|
Set the AES decryption initialization vector binary block from given hexadecimal representation. |
|
@end table |
|
|
|
Accepted URL formats: |
|
@example |
|
crypto:@var{URL} |
|
crypto+@var{URL} |
|
@end example |
|
|
|
@section data |
|
|
|
Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}. |
|
|
|
For example, to convert a GIF file given inline with @command{ffmpeg}: |
|
@example |
|
ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png |
|
@end example |
|
|
|
@section file |
|
|
|
File access protocol. |
|
|
|
Read from or write to a file. |
|
|
|
A file URL can have the form: |
|
@example |
|
file:@var{filename} |
|
@end example |
|
|
|
where @var{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 @file{input.mpeg} with @command{ffmpeg} |
|
use the command: |
|
@example |
|
ffmpeg -i file:input.mpeg output.mpeg |
|
@end example |
|
|
|
This protocol accepts the following options: |
|
|
|
@table @option |
|
@item truncate |
|
Truncate existing files on write, if set to 1. A value of 0 prevents |
|
truncating. Default value is 1. |
|
|
|
@item blocksize |
|
Set I/O operation maximum block size, in bytes. Default value is |
|
@code{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. |
|
@end table |
|
|
|
@section ftp |
|
|
|
FTP (File Transfer Protocol). |
|
|
|
Read from or write to remote resources using FTP protocol. |
|
|
|
Following syntax is required. |
|
@example |
|
ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg |
|
@end example |
|
|
|
This protocol accepts the following options. |
|
|
|
@table @option |
|
@item 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 ftp-anonymous-password |
|
Password used when login as anonymous user. Typically an e-mail address |
|
should be used. |
|
|
|
@item 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. |
|
@end table |
|
|
|
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. |
|
|
|
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 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 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 or for listen mode. |
|
|
|
@item http_proxy |
|
set HTTP proxy to tunnel through e.g. http://example.com:1234 |
|
|
|
@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 the protocol will use a |
|
string describing the libavformat build. ("Lavf/<version>") |
|
|
|
@item user-agent |
|
This is a deprecated option, you can use user_agent instead it. |
|
|
|
@item 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 reconnect_at_eof |
|
If set then eof is treated like an error and causes reconnection, this is useful |
|
for live / endless streams. |
|
|
|
@item reconnect_streamed |
|
If set then even streamed/non seekable streams will be reconnected on errors. |
|
|
|
@item reconnect_delay_max |
|
Sets the maximum delay in seconds after which to give up reconnecting |
|
|
|
@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 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 offset |
|
Set initial byte offset. |
|
|
|
@item end_offset |
|
Try to limit the request to bytes preceding this offset. |
|
|
|
@item 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 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 or ffserver.c and thus must not be used as a command line option. |
|
@example |
|
# Server side (sending): |
|
ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port} |
|
|
|
# Client side (receiving): |
|
ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg |
|
|
|
# Client can also be done with wget: |
|
wget http://@var{server}:@var{port} -O somefile.ogg |
|
|
|
# Server side (receiving): |
|
ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg |
|
|
|
# Client side (sending): |
|
ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port} |
|
|
|
# Client can also be done with wget: |
|
wget --post-file=somefile.ogg http://@var{server}:@var{port} |
|
@end example |
|
|
|
@end table |
|
|
|
@subsection HTTP Cookies |
|
|
|
Some HTTP requests will be denied unless cookie values are passed in with the |
|
request. The @option{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: |
|
@example |
|
ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8 |
|
@end example |
|
|
|
@section Icecast |
|
|
|
Icecast protocol (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. |
|
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 |
|
|
|
@example |
|
icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint} |
|
@end example |
|
|
|
@section mmst |
|
|
|
MMS (Microsoft Media Server) protocol over TCP. |
|
|
|
@section mmsh |
|
|
|
MMS (Microsoft Media Server) protocol over HTTP. |
|
|
|
The required syntax is: |
|
@example |
|
mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] |
|
@end example |
|
|
|
@section 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. |
|
@example |
|
# 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: |
|
@end example |
|
|
|
Note that some formats (typically MOV) require the output protocol to |
|
be seekable, so they will fail with the MD5 output protocol. |
|
|
|
@section pipe |
|
|
|
UNIX pipe access protocol. |
|
|
|
Read and write from UNIX pipes. |
|
|
|
The accepted syntax is: |
|
@example |
|
pipe:[@var{number}] |
|
@end example |
|
|
|
@var{number} is the number corresponding to the file descriptor of the |
|
pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{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 @command{ffmpeg}: |
|
@example |
|
cat test.wav | ffmpeg -i pipe:0 |
|
# ...this is the same as... |
|
cat test.wav | ffmpeg -i pipe: |
|
@end example |
|
|
|
For writing to stdout with @command{ffmpeg}: |
|
@example |
|
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 |
|
@end example |
|
|
|
This protocol accepts the following options: |
|
|
|
@table @option |
|
@item blocksize |
|
Set I/O operation maximum block size, in bytes. Default value is |
|
@code{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. |
|
@end table |
|
|
|
Note that some formats (typically MOV), require the output protocol to |
|
be seekable, so they will fail with the pipe output protocol. |
|
|
|
@section 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 @code{rtp_mpegts} muxer and |
|
the @code{rtp} protocol. |
|
|
|
The required syntax is: |
|
@example |
|
-f rtp_mpegts -fec prompeg=@var{option}=@var{val}... rtp://@var{hostname}:@var{port} |
|
@end example |
|
|
|
The destination UDP ports are @code{port + 2} for the column FEC stream |
|
and @code{port + 4} for the row FEC stream. |
|
|
|
This protocol accepts the following options: |
|
@table @option |
|
|
|
@item l=@var{n} |
|
The number of columns (4-20, LxD <= 100) |
|
|
|
@item d=@var{n} |
|
The number of rows (4-20, LxD <= 100) |
|
|
|
@end table |
|
|
|
Example usage: |
|
|
|
@example |
|
-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://@var{hostname}:@var{port} |
|
@end example |
|
|
|
@section 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: |
|
@example |
|
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 port |
|
The number of the TCP port to use (by default is 1935). |
|
|
|
@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.). 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:". 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 |
|
|
|
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{ffplay} a multimedia resource named |
|
"sample" from the application "vod" from an RTMP server "myserver": |
|
@example |
|
ffplay rtmp://myserver/vod/sample |
|
@end example |
|
|
|
To publish to a password protected server, passing the playpath and |
|
app names separately: |
|
@example |
|
ffmpeg -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 libsmbclient |
|
|
|
libsmbclient permits one to manipulate CIFS/SMB network resources. |
|
|
|
Following syntax is required. |
|
|
|
@example |
|
smb://[[domain:]user[:password@@]]server[/share[/path[/file]]] |
|
@end example |
|
|
|
This protocol accepts the following options. |
|
|
|
@table @option |
|
@item 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. |
|
|
|
@item truncate |
|
Truncate existing files on write, if set to 1. A value of 0 prevents |
|
truncating. Default value is 1. |
|
|
|
@item workgroup |
|
Set the workgroup used for making connections. By default workgroup is not specified. |
|
|
|
@end table |
|
|
|
For more information see: @url{http://www.samba.org/}. |
|
|
|
@section libssh |
|
|
|
Secure File Transfer Protocol via libssh |
|
|
|
Read from or write to remote resources using SFTP protocol. |
|
|
|
Following syntax is required. |
|
|
|
@example |
|
sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg |
|
@end example |
|
|
|
This protocol accepts the following options. |
|
|
|
@table @option |
|
@item 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 truncate |
|
Truncate existing files on write, if set to 1. A value of 0 prevents |
|
truncating. Default value is 1. |
|
|
|
@item private_key |
|
Specify the path of the file containing private key to use during authorization. |
|
By default libssh searches for keys in the @file{~/.ssh/} directory. |
|
|
|
@end table |
|
|
|
Example: Play a file stored on remote server. |
|
|
|
@example |
|
ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg |
|
@end example |
|
|
|
@section 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: |
|
@example |
|
@var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options} |
|
@end example |
|
|
|
where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe", |
|
"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and |
|
@var{server}, @var{port}, @var{app} and @var{playpath} have the same |
|
meaning as specified for the RTMP native protocol. |
|
@var{options} contains a list of space-separated options of the form |
|
@var{key}=@var{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 |
|
@command{ffmpeg}: |
|
@example |
|
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream |
|
@end example |
|
|
|
To play the same stream using @command{ffplay}: |
|
@example |
|
ffplay "rtmp://myserver/live/mystream live=1" |
|
@end example |
|
|
|
@section rtp |
|
|
|
Real-time Transport Protocol. |
|
|
|
The required syntax for an RTP URL is: |
|
rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...] |
|
|
|
@var{port} specifies the RTP port to use. |
|
|
|
The following URL options are supported: |
|
|
|
@table @option |
|
|
|
@item ttl=@var{n} |
|
Set the TTL (Time-To-Live) value (for multicast only). |
|
|
|
@item rtcpport=@var{n} |
|
Set the remote RTCP port to @var{n}. |
|
|
|
@item localrtpport=@var{n} |
|
Set the local RTP port to @var{n}. |
|
|
|
@item localrtcpport=@var{n}' |
|
Set the local RTCP port to @var{n}. |
|
|
|
@item pkt_size=@var{n} |
|
Set max packet size (in bytes) to @var{n}. |
|
|
|
@item connect=0|1 |
|
Do a @code{connect()} on the UDP socket (if set to 1) or not (if set |
|
to 0). |
|
|
|
@item sources=@var{ip}[,@var{ip}] |
|
List allowed source IP addresses. |
|
|
|
@item block=@var{ip}[,@var{ip}] |
|
List disallowed (blocked) source IP addresses. |
|
|
|
@item 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 localport=@var{n} |
|
Set the local RTP port to @var{n}. |
|
|
|
This is a deprecated option. Instead, @option{localrtpport} should be |
|
used. |
|
|
|
@end table |
|
|
|
Important notes: |
|
|
|
@enumerate |
|
|
|
@item |
|
If @option{rtcpport} is not set the RTCP port will be set to the RTP |
|
port value plus 1. |
|
|
|
@item |
|
If @option{localrtpport} (the local RTP port) is not set any available |
|
port will be used for the local RTP and RTCP ports. |
|
|
|
@item |
|
If @option{localrtcpport} (the local RTCP port) is not set it will be |
|
set to the local RTP port value plus 1. |
|
@end enumerate |
|
|
|
@section 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 |
|
@uref{https://github.com/revmischa/rtsp-server, RTSP server}). |
|
|
|
The required syntax for a RTSP url is: |
|
@example |
|
rtsp://@var{hostname}[:@var{port}]/@var{path} |
|
@end example |
|
|
|
Options can be set on the @command{ffmpeg}/@command{ffplay} command |
|
line, or set in code via @code{AVOption}s or in |
|
@code{avformat_open_input}. |
|
|
|
The following options are supported. |
|
|
|
@table @option |
|
@item initial_pause |
|
Do not start playing the stream immediately if set to 1. Default value |
|
is 0. |
|
|
|
@item rtsp_transport |
|
Set RTSP transport protocols. |
|
|
|
It accepts the following values: |
|
@table @samp |
|
@item udp |
|
Use UDP as lower transport protocol. |
|
|
|
@item tcp |
|
Use TCP (interleaving within the RTSP control channel) as lower |
|
transport protocol. |
|
|
|
@item udp_multicast |
|
Use UDP multicast as lower transport protocol. |
|
|
|
@item http |
|
Use HTTP tunneling as lower transport protocol, which is useful for |
|
passing proxies. |
|
@end table |
|
|
|
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 @samp{tcp} and @samp{udp} options are supported. |
|
|
|
@item rtsp_flags |
|
Set RTSP flags. |
|
|
|
The following values are accepted: |
|
@table @samp |
|
@item filter_src |
|
Accept packets only from negotiated peer address and port. |
|
@item listen |
|
Act as a server, listening for an incoming connection. |
|
@item prefer_tcp |
|
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport. |
|
@end table |
|
|
|
Default value is @samp{none}. |
|
|
|
@item allowed_media_types |
|
Set media types to accept from the server. |
|
|
|
The following flags are accepted: |
|
@table @samp |
|
@item video |
|
@item audio |
|
@item data |
|
@end table |
|
|
|
By default it accepts all media types. |
|
|
|
@item min_port |
|
Set minimum local UDP port. Default value is 5000. |
|
|
|
@item max_port |
|
Set maximum local UDP port. Default value is 65000. |
|
|
|
@item timeout |
|
Set maximum timeout (in seconds) to wait for incoming connections. |
|
|
|
A value of -1 means infinite (default). This option implies the |
|
@option{rtsp_flags} set to @samp{listen}. |
|
|
|
@item reorder_queue_size |
|
Set number of packets to buffer for handling of reordered packets. |
|
|
|
@item stimeout |
|
Set socket TCP I/O timeout in microseconds. |
|
|
|
@item user-agent |
|
Override User-Agent header. If not specified, it defaults to the |
|
libavformat identifier string. |
|
@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). 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 @command{ffplay}, 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}. |
|
|
|
@subsection Examples |
|
|
|
The following examples all make use of the @command{ffplay} and |
|
@command{ffmpeg} tools. |
|
|
|
@itemize |
|
@item |
|
Watch a stream over UDP, with a max reordering delay of 0.5 seconds: |
|
@example |
|
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 |
|
@end example |
|
|
|
@item |
|
Watch a stream tunneled over HTTP: |
|
@example |
|
ffplay -rtsp_transport http rtsp://server/video.mp4 |
|
@end example |
|
|
|
@item |
|
Send a stream in realtime to a RTSP server, for others to watch: |
|
@example |
|
ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp |
|
@end example |
|
|
|
@item |
|
Receive a stream in realtime: |
|
@example |
|
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output} |
|
@end example |
|
@end itemize |
|
|
|
@section 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. |
|
|
|
@subsection Muxer |
|
|
|
The syntax for a SAP url given to the muxer is: |
|
@example |
|
sap://@var{destination}[:@var{port}][?@var{options}] |
|
@end example |
|
|
|
The RTP packets are sent to @var{destination} on port @var{port}, |
|
or to port 5004 if no port is specified. |
|
@var{options} is a @code{&}-separated list. The following options |
|
are supported: |
|
|
|
@table @option |
|
|
|
@item announce_addr=@var{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 @var{destination} is an IPv6 address. |
|
|
|
@item announce_port=@var{port} |
|
Specify the port to send the announcements on, defaults to |
|
9875 if not specified. |
|
|
|
@item ttl=@var{ttl} |
|
Specify the time to live value for the announcements and RTP packets, |
|
defaults to 255. |
|
|
|
@item same_port=@var{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. |
|
@end table |
|
|
|
Example command lines follow. |
|
|
|
To broadcast a stream on the local subnet, for watching in VLC: |
|
|
|
@example |
|
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1 |
|
@end example |
|
|
|
Similarly, for watching in @command{ffplay}: |
|
|
|
@example |
|
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255 |
|
@end example |
|
|
|
And for watching in @command{ffplay}, over IPv6: |
|
|
|
@example |
|
ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4] |
|
@end example |
|
|
|
@subsection Demuxer |
|
|
|
The syntax for a SAP url given to the demuxer is: |
|
@example |
|
sap://[@var{address}][:@var{port}] |
|
@end example |
|
|
|
@var{address} is the multicast address to listen for announcements on, |
|
if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{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: |
|
|
|
@example |
|
ffplay sap:// |
|
@end example |
|
|
|
To play back the first stream announced on one the default IPv6 SAP multicast address: |
|
|
|
@example |
|
ffplay sap://[ff0e::2:7ffe] |
|
@end example |
|
|
|
@section sctp |
|
|
|
Stream Control Transmission Protocol. |
|
|
|
The accepted URL syntax is: |
|
@example |
|
sctp://@var{host}:@var{port}[?@var{options}] |
|
@end example |
|
|
|
The protocol accepts the following options: |
|
@table @option |
|
@item listen |
|
If set to any value, listen for an incoming connection. Outgoing connection is done by default. |
|
|
|
@item max_streams |
|
Set the maximum number of streams. By default no limit is set. |
|
@end table |
|
|
|
@section srtp |
|
|
|
Secure Real-time Transport Protocol. |
|
|
|
The accepted options are: |
|
@table @option |
|
@item srtp_in_suite |
|
@item srtp_out_suite |
|
Select input and output encoding suites. |
|
|
|
Supported values: |
|
@table @samp |
|
@item AES_CM_128_HMAC_SHA1_80 |
|
@item SRTP_AES128_CM_HMAC_SHA1_80 |
|
@item AES_CM_128_HMAC_SHA1_32 |
|
@item SRTP_AES128_CM_HMAC_SHA1_32 |
|
@end table |
|
|
|
@item srtp_in_params |
|
@item 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. |
|
@end table |
|
|
|
@section subfile |
|
|
|
Virtually extract a segment of a file or another stream. |
|
The underlying stream must be seekable. |
|
|
|
Accepted options: |
|
@table @option |
|
@item start |
|
Start offset of the extracted segment, in bytes. |
|
@item end |
|
End offset of the extracted segment, in bytes. |
|
If set to 0, extract till end of file. |
|
@end table |
|
|
|
Examples: |
|
|
|
Extract a chapter from a DVD VOB file (start and end sectors obtained |
|
externally and multiplied by 2048): |
|
@example |
|
subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB |
|
@end example |
|
|
|
Play an AVI file directly from a TAR archive: |
|
@example |
|
subfile,,start,183241728,end,366490624,,:archive.tar |
|
@end example |
|
|
|
Play a MPEG-TS file from start offset till end: |
|
@example |
|
subfile,,start,32815239,end,0,,:video.ts |
|
@end example |
|
|
|
@section tee |
|
|
|
Writes the output to multiple protocols. The individual outputs are separated |
|
by | |
|
|
|
@example |
|
tee:file://path/to/local/this.avi|file://path/to/local/that.avi |
|
@end example |
|
|
|
@section tcp |
|
|
|
Transmission Control Protocol. |
|
|
|
The required syntax for a TCP url is: |
|
@example |
|
tcp://@var{hostname}:@var{port}[?@var{options}] |
|
@end example |
|
|
|
@var{options} contains a list of &-separated options of the form |
|
@var{key}=@var{val}. |
|
|
|
The list of supported options follows. |
|
|
|
@table @option |
|
@item listen=@var{1|0} |
|
Listen for an incoming connection. Default value is 0. |
|
|
|
@item timeout=@var{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 listen_timeout=@var{milliseconds} |
|
Set listen timeout, expressed in milliseconds. |
|
|
|
@item recv_buffer_size=@var{bytes} |
|
Set receive buffer size, expressed bytes. |
|
|
|
@item send_buffer_size=@var{bytes} |
|
Set send buffer size, expressed bytes. |
|
|
|
@item tcp_nodelay=@var{1|0} |
|
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0. |
|
@end table |
|
|
|
The following example shows how to setup a listening TCP connection |
|
with @command{ffmpeg}, which is then accessed with @command{ffplay}: |
|
@example |
|
ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen |
|
ffplay tcp://@var{hostname}:@var{port} |
|
@end example |
|
|
|
@section tls |
|
|
|
Transport Layer Security (TLS) / Secure Sockets Layer (SSL) |
|
|
|
The required syntax for a TLS/SSL url is: |
|
@example |
|
tls://@var{hostname}:@var{port}[?@var{options}] |
|
@end example |
|
|
|
The following parameters can be set via command line options |
|
(or in code via @code{AVOption}s): |
|
|
|
@table @option |
|
|
|
@item ca_file, cafile=@var{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 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, cert=@var{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 key_file, key=@var{filename} |
|
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 |
|
|
|
Example command lines: |
|
|
|
To create a TLS/SSL server that serves an input stream. |
|
|
|
@example |
|
ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key} |
|
@end example |
|
|
|
To play back a stream from the TLS/SSL server using @command{ffplay}: |
|
|
|
@example |
|
ffplay tls://@var{hostname}:@var{port} |
|
@end example |
|
|
|
@section udp |
|
|
|
User Datagram Protocol. |
|
|
|
The required syntax for an UDP URL is: |
|
@example |
|
udp://@var{hostname}:@var{port}[?@var{options}] |
|
@end example |
|
|
|
@var{options} contains a list of &-separated options of the form @var{key}=@var{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 @var{fifo_size} and |
|
@var{overrun_nonfatal} options are related to this buffer. |
|
|
|
The list of supported options follows. |
|
|
|
@table @option |
|
@item buffer_size=@var{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 @var{fifo_size}. |
|
|
|
@item bitrate=@var{bitrate} |
|
If set to nonzero, the output will have the specified constant bitrate if the |
|
input has enough packets to sustain it. |
|
|
|
@item burst_bits=@var{bits} |
|
When using @var{bitrate} this specifies the maximum number of bits in |
|
packet bursts. |
|
|
|
@item localport=@var{port} |
|
Override the local UDP port to bind with. |
|
|
|
@item localaddr=@var{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 pkt_size=@var{size} |
|
Set the size in bytes of UDP packets. |
|
|
|
@item reuse=@var{1|0} |
|
Explicitly allow or disallow reusing UDP sockets. |
|
|
|
@item ttl=@var{ttl} |
|
Set the time to live value (for multicast only). |
|
|
|
@item connect=@var{1|0} |
|
Initialize the UDP socket with @code{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 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. |
|
|
|
@item fifo_size=@var{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 overrun_nonfatal=@var{1|0} |
|
Survive in case of UDP receiving circular buffer overrun. Default |
|
value is 0. |
|
|
|
@item timeout=@var{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 broadcast=@var{1|0} |
|
Explicitly allow or disallow UDP broadcasting. |
|
|
|
Note that broadcasting may not work properly on networks having |
|
a broadcast storm protection. |
|
@end table |
|
|
|
@subsection Examples |
|
|
|
@itemize |
|
@item |
|
Use @command{ffmpeg} to stream over UDP to a remote endpoint: |
|
@example |
|
ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port} |
|
@end example |
|
|
|
@item |
|
Use @command{ffmpeg} to stream in mpegts format over UDP using 188 |
|
sized UDP packets, using a large input buffer: |
|
@example |
|
ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535 |
|
@end example |
|
|
|
@item |
|
Use @command{ffmpeg} to receive over UDP from a remote endpoint: |
|
@example |
|
ffmpeg -i udp://[@var{multicast-address}]:@var{port} ... |
|
@end example |
|
@end itemize |
|
|
|
@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
|
|
|