This patch adds support for:
- ffplay ipfs://<cid>
- ffplay ipns://<cid>
IPFS data can be played from so called "ipfs gateways".
A gateway is essentially a webserver that gives access to the
distributed IPFS network.
This protocol support (ipfs and ipns) therefore translates
ipfs:// and ipns:// to a http:// url. This resulting url is
then handled by the http protocol. It could also be https
depending on the gateway provided.
To use this protocol, a gateway must be provided.
If you do nothing it will try to find it in your
$HOME/.ipfs/gateway file. The ways to set it manually are:
1. Define a -gateway <url> to the gateway.
2. Define $IPFS_GATEWAY with the full http link to the gateway.
3. Define $IPFS_PATH and point it to the IPFS data path.
4. Have IPFS running in your local user folder (under $HOME/.ipfs).
Signed-off-by: Mark Gaiser <markg85@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
And forward it to the underlying UDP protocol.
Fixes ticket #7517.
Signed-off-by: Jiangjie Gao <gaojiangjie@live.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Add the "http_proxy" option and its handling to the "tls" protocol,
pass the option from the "https" protocol.
The "https" protocol already defines the "http_proxy" command line
option, like the "http" protocol does. The "http" protocol properly
honors that command line option in addition to the environment
variable. The "https" protocol doesn't, because the proxy is
evaluated in the underlying "tls" protocol, which doesn't have this
option, and thus only handles the environment variable, which it
has access to.
Fixes#7223.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Marton Balint <cus@passwd.hu>
Also remove AV_LOG_SIMULATE from the list as it is not used directly, and do
not use panic level on unknown loglevel, but make them warn. Also fix mapping of
NOTICE/INFO/VERBOSE and add documentation about when the option should actually
be used.
Signed-off-by: Marton Balint <cus@passwd.hu>
Maximum packet size is 10000 (RIST_MAX_PACKET_SIZE, which is unfortunately
private) minus the RIST protocol overhead which is 28 bytes for the unencrypted
case, 36 for the encrypted case.
Signed-off-by: Marton Balint <cus@passwd.hu>
This commit adds a "gophers" handler to the gopher protocol. gophers
is a community-adopted protocol that acts the same way like normal
gopher with the added TLS encapsulation.
The gophers protocol is supported by gopher servers like geomydae(8),
and clients like curl(1), clic(1), and hurl(1).
This commit also adds compilation guards to both gopher and gophers,
since now there are two protocols in the file it makes sense to
have this addition.
Signed-off-by: parazyd <parazyd@dyne.org>
Signed-off-by: Marton Balint <cus@passwd.hu>
Add 2 new options:
- reconnect_on_http_error - a list of http status codes that should be
retried. the list can contain explicit status codes / the strings
4xx/5xx.
- reconnect_on_network_error - reconnects on arbitrary errors during
connect, e.g. ECONNRESET/ETIMEDOUT
the retry employs the same exponential backoff logic as the existing
reconnect/reconnect_at_eof flags.
related tickets:
https://trac.ffmpeg.org/ticket/6066https://trac.ffmpeg.org/ticket/7768
Signed-off-by: Marton Balint <cus@passwd.hu>
then we can set the rtp read timeout instead of infinite timeout.
How to test(5s timeout):
./ffprobe -i rtp://192.168.1.67:1234?timeout=5000000
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
remove the timeout option docs part for HTTP protocol and add
auth_type option part.
Reviewed-by: Gyan Doshi <ffmpeg@gyani.pro>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
Supports connecting to a RabbitMQ broker via AMQP version 0-9-1.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
rw_timeout is the generic URLcontext option, not the protocol specific timeout
option, also ?rw_timeout never worked because ?timeout was parsed instead.
Signed-off-by: Marton Balint <cus@passwd.hu>
It is a common mistake that people only increase fifo_size when they experience
drops, unfortunately this does not help for higher bitrate (> 100 Mbps) streams
when the reader thread simply might not receive the packets in time (especially
under high CPU load) if the default 64 KB of kernel buffer size is used.
New default is determined so that common linux systems can set this buffer size
without tuning kernel parameters.
Signed-off-by: Marton Balint <cus@passwd.hu>
add linger parameter to libsrt, it's setting the number of seconds
that the socket waits for unsent data when closing.
Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
This introduces two new AVOption options for the FTP protocol,
one named ftp-user to supply the username to be used for auth,
one named ftp-password to supply the password to be used for auth.
These are useful for when an API user does not wish to deal with
URL manipulation and percent encoding.
Setting them while also having credentials in the URL will use the
credentials from the URL. The rationale for this is that credentials
embedded in the URL are probably more specific to what the user is
trying to do than anything set by some API user.
Signed-off-by: Nicolas Frattaroli <ffmpeg@fratti.ch>
Signed-off-by: Marton Balint <cus@passwd.hu>
When ffmpeg was streaming, multiple clients were only supported by using a
multicast destination address. An alternative was to stream to a server which
re-distributes the content. This commit adds ZeroMQ as a protocol, which allows
multiple clients to connect to a single ffmpeg instance.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fix ticket #7297
The current setting for send-expect-100 option is either
enabled if applicable or forced enabled, no option to force
disable the header. This change is to expand the option setting
to provide more flexibility, which is useful for rstp case.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Several SRT options are missing. Since pkg_config requires libsrt v1.3.0 and above, it should be able to support options added in libsrt v1.3.0 and below.
This commit adds 8 SRT options.
sndbuf, rcvbuf, lossmaxttl, minversion, streamid, smoother, messageapi and transtype
The keys of option are equivalent to stransmit.
https://github.com/Haivision/srt/blob/v1.3.0/apps/socketoptions.hpp#L196-L223
Signed-off-by: Marton Balint <cus@passwd.hu>
This allows getting data only from a specific source IP. This is useful not
only for unicast but for multicast as well because multicast source
subscriptions do not act as source filters for the incoming packets.
Signed-off-by: Marton Balint <cus@passwd.hu>
Also make sure we set the URL context max packet size accordingly.
Based on a patch by Tudor Suciu <tudor.suciu@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
The protocol requires libsrt (https://github.com/Haivision/srt) to be
installed
Signed-off-by: Sven Dueking <sven.dueking@nablet.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Can be used by the api user to figure out what http features the server supports based on the response received.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Mark Gaiser
Limin Wang
James Almer
Thilo Borgmann
Nick Ruff
Zhao Zhili
Zhao Zhili
Andreas Rheinhardt
Andriy Gelman
Jiangjie Gao
Moritz Barsnick
Marton Balint
Paul B Mahol
parazyd
Lingjiang Fang
erankor
Gyan Doshi
Jun Zhao
Marvin Scholz
Levis Florian
Nicolas Frattaroli
Jun Li
Matsuzawa Tomohiro
Jun Zhao
Sven Dueking
Steven Liu
Rostislav Pehlivanov
Aman Gupta