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Author SHA1 Message Date
Mathieu Duponchelle
a455819871 webrtcsink: fix tracking of signaller state 
For the signaller to get stopped, we need to remember that we started it
in the first place.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
3368f55a88 webrtcsink: don't return value from error closure 
the signal doesn't expect a return value, which meant we were panicking
as soon as the signaller tried to report an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
58c8c0edc7 webrtc: signaller iface: fix session-ended vs end-session confusion 
Session ending is bidirectional: the signaller can tell the sink that a
session was ended, and the sink can tell the signaller to end a session.

As such, two signals are needed, before this patch the second case was
not working as in essence the sink was telling itself that a session was
ended, and obviously failing to even find it when trying to end it again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Tim-Philipp Müller
7c30430320 webrtc-api: replace LICENSE file symlink with copy 
As in !1157

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1169>
 2023-04-08 17:22:37 +01:00
Matthew Waters
e69b4b7f45 webrtc/signaller/iface: give variables appropriate names 
Rather than arg0, arg1, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
4f4e5f0d75 webrtcsink/signaller: don't call signals while having state/settings locked 
It is a recipe for deadlocks if the signal callback calls back into
webrtcsink in some way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
1c61e46f37 webrtcsink: privatise signalling functions 
The functionality is now access through the relevant signals instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
2ac560975c webrtc/signaller: emit the relevant signals instead of the interface vtable 
In order to support the use case of an external user providing their own
signalling mechanism, we want the signals to be used and only if nothing
is connected, fallback to the default handling.  Calling the interface
vtable directly will bypass the signal emission entirely.

Also ensure that the signals are defined properly for this case. i.e.
1. Signals the the application/external code is expected to emit are
   marked as an action signal.
2. Add accumulators to avoid calling the default class handler if
   another signal handler is connected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
343b659755 webrtc/signaller: remove SignallableImplExt 
This pattern is used for subclassing and calling parent class/interface functions.
However that is not useful for the signaller object.
1. The signals are the API contract and should instead be used by
   webrtcsrc/sink to ask or provide outside for/with information.
2. The default case (no signal attached)is instead handled by default class
   handlers that call directly using the relevant rust trait.  No parent
   (GObject) vfuncs necessary.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
b6e78b5f04 webrtcsink: expose signaller as a property 
in the process move the signaller field to the settings struct

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Thibault Saunier
8236f3e5e7 webrtcsink: Port to the 'webrtcsrc' signaller object/interface 
With contributions from:
Matthew Waters <matthew@centricular.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:03:47 +10:00
Loïc Le Page
f17622a1e1 webrtc: Add gstwebrtc-api subproject in net/webrtc plugin 
This subproject adds a high-level web API compatible with GStreamer
webrtcsrc and webrtcsink elements and the corresponding signaling
server. It allows a perfect bidirectional communication between HTML5
WebRTC API and native GStreamer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/946>
 2023-04-04 16:29:44 +02:00
Tim-Philipp Müller
8845f6a4c6 git: replace LICENSE file symlinks with copies 
Git will de-duplicate the contents for us anyway, and
symlinks can cause problems with some versions of git
and also on Windows.

https://github.com/mesonbuild/meson/issues/11646
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4326

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1157>
 2023-04-04 14:26:37 +01:00
Mathieu Duponchelle
15e1844956 webrtcsink: fix calculation of fec_ratio with multiple encoders 
In this context, the bitrate variable is for all encoders, but the
max_bitrate field is per encoder. To calculate a proper FEC ratio, we
need to scale max_bitrate to the number of encoders.

+ Also clamp the fec-percentage that we set on the transceiver for extra
  safety

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1151>
 2023-03-31 12:19:07 +00:00
Sebastian Dröge
315e53f064 webrtc: Update to AWS SDK 0.55/0.25 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1152>
 2023-03-31 09:12:26 +00:00
David Revay
002a70a2a4 chore(webrtcsink): fix max-bitrate blurb and nick 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1150>
 2023-03-28 16:11:05 +11:00
Vivia Nikolaidou
7a1b2d97d4 webrtcsink: Add ice-transport-policy option 
Can be used to force relay ICE candidates, ensuring TURN server is used.
Proxy to the corresponding setting in webrtcbin,

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1143>
 2023-03-27 16:12:13 +03:00
Sebastian Dröge
c1bac30694 webrtc: Update to aws 0.54/0.24 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1131>
 2023-03-11 09:37:14 +02:00
Mathieu Duponchelle
584392049c net/webrtc: implement AWS KVS signaller 
And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink.

This adds support in webrtcsink for processing a consumer offer, instead
of producing one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
 2023-03-09 15:39:09 +00:00
Sebastian Dröge
fc5ed15af5 Update for gst::Element::link_many() and related API generalization 
Specifically, get rid of now unneeded `&`.
 2023-03-09 16:46:52 +02:00
Thibault Saunier
ce3bb2f1d4 Add a webrtcsrc element 
Updating the docker image to include:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3236

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
 2023-02-28 20:50:15 -03:00
Thibault Saunier
0ae637f531 webrtcsink: Move RUNTIME to the crate so it can be reused 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
 2023-02-28 17:57:14 -03:00
Thibault Saunier
4ec441560b webrtc: Enhance debug messages when using unknown peer ID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
 2023-02-28 19:28:51 +00:00
Matthew Waters
542c7e12b8 webrtcsink: also support nvvidconv in lieu of nvvideoconvert 
nvvideoconvert may not exist and nvvidconv might on some Jetson
platforms.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1107>
 2023-02-28 10:12:36 +11:00
Sebastian Dröge
9fc1404415 Update minimum supported Rust version to 1.66 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
 2023-02-20 11:09:01 +02:00
Sebastian Dröge
ac8afc4ac0 Update to async-tungstenite 0.20 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1087>
 2023-02-10 13:03:07 +02:00
Sebastian Dröge
1e13dbb99c Update versions to 0.11.0-alpha.1 2023-02-10 00:23:56 +02:00
Sebastian Dröge
560bdc4cb7 Update for glib API changes 2023-01-31 12:24:07 +02:00
Sebastian Dröge
3b4c48d9f5 Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1062>
 2023-01-25 10:31:19 +02:00
Sebastian Dröge
2c386fb792 Update for various deprecated APIs 2023-01-22 20:07:26 +02:00
Sebastian Dröge
4582ae91ab Move remaining plugins to ParamSpec builders 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1054>
 2023-01-21 18:34:55 +02:00
Sebastian Dröge
812df78b75 webrtcbin: Update for StreamProducer API changes 2023-01-16 16:36:41 +02:00
Sebastian Dröge
6132788b02 Update for caps/structure-related string API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1048>
 2023-01-15 22:58:44 +02:00
Mathieu Duponchelle
1a8abde884 webrtcsink: fix panic on pre-bwe request error 
We dispose of consumer pipelines asynchronously, potentially after the
session objects have been disposed of.

As session objects are the owner of the cc element, it is entirely
possible for the bwe-request signal to get emitted after cc has been
disposed of, as the closure only takes a weak reference to it.

Fix by simply checking if cc is None

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1044>
 2023-01-11 15:09:45 +00:00
Sebastian Dröge
27435ad82e Update for API changes 2023-01-05 12:33:54 +02:00
Zhao, Gang
9fa838e366 webrtc: Fix rustfmt errors 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-27 11:12:54 +02:00
Zhao, Gang
877a9bd7f3 webrtc: Share runtime between webrtcsink and signaller crates 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00
Zhao, Gang
1ffeb4d44d webrtc: Move from async-std to tokio 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00
Zhao, Gang
2bc29c1fd3 webrtc: examples: Update package-lock.json 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00
Mathieu Duponchelle
e5360ff431 webrtc/README: update command to run the signalling server 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/277

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1012>
 2022-12-13 12:47:26 +01:00
Sebastian Dröge
3f904553ea Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1011>
 2022-12-13 11:43:16 +02:00
Sebastian Dröge
fb42cd8a0f net: Update to async-tungstenite 0.19 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1005>
 2022-12-11 12:54:24 +02:00
Raphael Dürscheid
aa2abc50bf webrtcsink: Support nvv4l2vp9enc 
Naive support for nvv4l2vp9enc by assuming configuration is equivalent
to existing nvv4l2vp8enc. Validated to have relevant properties.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/983>
 2022-12-02 10:18:27 +00:00
Sebastian Dröge
fceacf7081 Update for gst::Array / gst::List API improvements 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/985>
 2022-11-27 01:12:46 +02:00
Thibault Saunier
6b11284e8a webrtcsink: Make the turn-server prop a turn-servers list 
So that we can simply specify several turn servers at once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/973>
 2022-11-16 14:48:16 +00:00
Guillaume Desmottes
37cb636140 webrtc: README: fix couple of links 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/975>
 2022-11-11 14:51:46 +01:00
Mathieu Duponchelle
66e7b314f7 webrtcsink: improve debug 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/972>
 2022-11-10 15:00:19 +00:00
Sebastian Dröge
a5f3197651 Add missing doc features to WebRTC plugins 2022-11-07 18:06:29 +00:00
Sebastian Dröge
a8250abbf1 Fix various new clippy warnings 2022-11-01 10:27:48 +02:00
Sebastian Dröge
976ae5707e webrtc: Update to human_bytes 0.4 2022-10-31 14:11:29 +02:00