GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI]
Features:
* Live streaming N audio and N video
- With RTCP-based A/V sync
* Lower transports: TCP, UDP, UDP-Multicast
* RTP, RTCP SR, RTCP RR
* OPTIONS DESCRIBE SETUP PLAY TEARDOWN
* Custom UDP socket management, does not use udpsrc/udpsink
* Supports both rtpbin and the rtpbin2 rust rewrite
- Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs)
* Properties:
- protocols selection and priority (NEW!)
- location supports rtsp[ut]://
- port-start instead of port-range
Co-Authored-by: Tim-Philipp Müller <tim@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
This new plugin exposes two elements, intersink and intersrc. These act
as wormholes for data in the same process and can be used to forward
data from one pipeline to another.
The implementation makes use of gstreamer-utils' StreamProducer, and
supports dynamically adding and removing consumers, before and after
producers, and changing producer names while PLAYING, both on the sink
and the src.
This initial implementation comes with a small demo, and a few tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1257>
Instead of having a matrix of jobs, use a single job running
all the tests like the linux jobs do.
This helps with improved cache hits as most of the deps are shared
between the builds.
In gst-rs we build each crate on its own, since not all
crates share the same features and some conflict with each other.
However currently, that isn't the case in plugins-rs and instead
we can be building all the crates with the same flags and simplify
the the script.
Close#241
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410
Created a new plugin 'webrtchttp' to implement all the
WebRTC HTTP protocols under /net/webrtc-http directory.
WhipSink wraps around 'webrtcbin' with HTTP capabilites
to exchange SDP offer/answer so an ICE/DTLS session can
be established between the encoder/media producer (WHIP client)
and the broadcasting ingestion endpoint (Media Server).
Once the ICE/DTLS session is set up, the media will
flow unidirectionally from the WHIP client to the
broadcasting ingestion endpoint (Media Server).
Spec:
https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
The element expects an array of "commands", as GstStructures,
in the form:
operation, pattern=<pattern>, ...
The only operation implemented for now is replace-all, eg:
replace-all, pattern=foo, replacement=bar
Other operations can be implemented if useful in the future,
eg. "match" could post a message to the bus when the pattern
is encountered.
The main use case for this is automatic speech recognition,
as implemented by eg awstranscribe as users may want to replace
swear words with tamer language.
Commands are applied in order.
The interface is usable through the CLI with the usual escaping
strategies, though trying to pass in actual regular expressions
through it is a bit tricky, as this introduces yet another
level of escaping.
This new crate consists of two elements, jsongstenc and jsongstparse
Both these elements can deal with an ndjson based format, consisting
for now of two item types: "Buffer" and "Header"
eg:
{"Header":{"format":"foobar"}}
{"Buffer":{"pts":0,"duration":43,"data":{"foo":"bar"}}}
jsongstparse will interpret this by first sending caps
application/x-json, format=foobar, then a buffer containing
{"foo":"bar"}, timestamped as required.
Elements further downstream can then interpret the data further.
jsongstenc will simply perform the reverse operation.
