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158 commits

Author SHA1 Message Date
Sebastian Dröge
3fc6220009 Update to AWS SDK 0.33 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1354>
 2023-10-09 11:28:05 +03:00
Taruntej Kanakamalla
245185d2f6 net/webrtc/whip_signaller: Use the correct URL during redirect 
Copy of 90e06dc3 for whipclientsink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1351>
 2023-10-06 13:11:46 +00:00
Maksym Khomenko
e4096b5157 webrtcsink: README: add documentation for custom signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1340>
 2023-10-06 12:58:04 +03:00
Maksym Khomenko
a9719cada2 webrtcsink: add custom signaller example 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1340>
 2023-10-06 12:58:03 +03:00
Sebastian Dröge
1c4833bc5d Update to AWS SDK 0.32 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1352>
 2023-10-06 09:11:17 +03:00
Sebastian Dröge
4569b7eca6 Fix various new 1.73 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1347>
 2023-10-03 17:47:30 +03:00
Piotr Brzeziński
fe4273ca2a webrtc: Fix paths in README 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1342>
 2023-09-29 17:05:29 +02:00
Sebastian Dröge
9595c6a1e5 Update to AWS SDK 0.31 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1334>
 2023-09-25 13:36:12 +03:00
Seungha Yang
225482f7ed webrtcsink: Propagate GstContext messages 
Implement CustomBusStream so that NEED_CONTEXT and HAVE_CONTEXT
messages from session/discovery can be forwarded to parent
pipeline and also GstContext can be shared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1322>
 2023-09-15 00:26:08 +09:00
Seungha Yang
1de7754616 webrtcsink: Add support for d3d11 memory and qsvh264enc 
Adding d3d11 memory and qsvh264enc support

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1322>
 2023-09-15 00:26:04 +09:00
Robert Ayrapetyan
18967dadbf gstwebrtc-api: drop guacamole 
fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/417

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1317>
 2023-09-11 19:21:41 +00:00
Mathieu Duponchelle
2381558169 webrtcsink: fix codec selection discoveries 
Since ab1ec12698:

webrtcsink: Add support for pre encoded streams

Discovery pipelines for remote offers were no longer fed any buffers.

While some encoders could already produce caps with no input buffers,
others, such as x264enc, simply hung forever. This resulted in no answer
getting produced if for instance video-caps were constrained to H264.

Fix this by tracking discovery pipelines at the State rather than the
InputStream level, removing the useless distinction of Initial vs.
CodecSelection discoveries, and always feeding all the current
discovery pipelines with incoming buffers.

For reference, the issue here was that codec selection discoveries were
assigned to local clones of InputStreams, not tracked anywhere, and thus
not iterated for discoveries when queuing incoming buffers from the
chain function, as it only looked at the original instance of
InputStream's in state.streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1319>
 2023-09-08 12:58:08 +00:00
Robert Ayrapetyan
e83238b681 webrtcsink: fix TWCC extension adding 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1310>
 2023-09-04 18:27:51 +00:00
Sebastian Dröge
8d433761d1 Fix indentation of let-else blocks 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1308>
 2023-08-29 12:21:38 +00:00
Taruntej Kanakamalla
de6d2e7f40 net/webrtc: rename whipwebrtcsink as whipclientsink 
add a deprecation warning in whipsink to indicate it
should be used only for RTP content
add documentation in whipsink code regarding usage and
deprecation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1282>
 2023-08-26 10:53:30 +05:30
Sebastian Dröge
905da44958 Update to AWS SDK 0.30 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1313>
 2023-08-25 09:46:52 +03:00
Andoni Morales Alastruey
3c1f05cdc3 webrtcsrc: document how to use the element for remote control 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
 2023-08-10 17:43:51 +00:00
Andoni Morales Alastruey
3000b08ec7 webrtcsrc: add support for navigation events 
This provides support GstNavigation events handling in webrtcsrc so that
a GStreamer client can be used to control remotely a GStreamer server,
similar to how the web client is capable of controlling a wpesrc.
This is part of a larger set of patches that require more work on the
sinks and sources.
server: d3d11screencapturesrc ! webrtcsink enable-data-channel-navigation=true
client: webrtcsrc enable-data-channel-navigation=true ! d3d11videosink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
 2023-08-10 17:43:51 +00:00
Loïc Le Page
e5e3dc6e19 net/webrtc/signaller: add property to get the connection client ID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
 2023-08-10 17:30:21 +02:00
Loïc Le Page
7af2ff0843 net/webrtc/signaller: advertise running producers in Listener mode 
When starting a webrtcsrc-signaller client in Listener mode, only the producers
started after the client connection were advertised. All currently
running producers were ignored unlike the gstwebrtc-api behavior. This
commit now lists all running producers when the client Listener connects
and advertises them through the "producer-added" signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
 2023-08-10 17:30:21 +02:00
Sebastian Dröge
d688aeb184 Update versions to 0.12.0-alpha.1 2023-08-10 17:21:11 +03:00
Sebastian Dröge
3b41f206bc Don't generate .def files for plugins 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
 2023-08-09 13:54:34 +03:00
Sebastian Dröge
b3826c108d webrtc: Update to async-tungstenite 0.23 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
 2023-08-09 13:18:44 +03:00
Sebastian Dröge
5ee46a214c webrtc: Use #[repr(C)] to get a C-compatible layout for the Signaller struct 
This is required by GObject for class/interface and instance structs and
the reason why implementing the `glib::ObjectInterface` trait is unsafe.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/397

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1297>
 2023-08-09 10:32:44 +03:00
Sebastian Dröge
cac791a6ca aws/webrtc: Update to AWS SDK 0.56/0.29 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1295>
 2023-08-07 20:03:51 +03:00
Sanchayan Maity
5b60ecbb18 net: webrtc/webrtchttp: Fix canceller usage 
Commit 08b6251a added the check to ensure only one canceller at a time for net/webrtc.

In `whipsink` and since `whipwebrtcsink` picked up the same implementation, there exists a
bug around the use of canceller. `whipsink` calls `wait_async` while passing the canceller
as an argument. The path `send_offer -> do_post -> parse_endpoint_response` results in the
canceller being replaced in each subsequent call to `wait_async`. Since `wait_async` call
does not ensure one canceller, with the async call the use of canceller/abort was subtly
broken. Similarly, for `whepsrc`.

We really don't need to use `wait_async` inside `do_post` for any `await` calls. If the
root future viz. `do_post` with `wait_async` is aborted, the child futures will be taken
care of.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1290>
 2023-08-04 10:01:11 +05:30
Mathieu Duponchelle
9680805bdb webrtcsink: don't forget to setup encoders for discoveries 
The "encoder-setup" signal must also be emitted for the encoders
used in discovery pipelines in order for the default settings to
be applied.

This otherwise meant that for instance the x264 encoder would
use a 60 frames latency, greatly delaying startup.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1289>
 2023-08-01 00:28:52 +02:00
Mathieu Duponchelle
dbeb65da06 webrtc/utils: fix typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1289>
 2023-08-01 00:28:32 +02:00
Sebastian Dröge
d4b3827efa webrtcsink: NVIDIA V4L2 encoders always require NVMM memory 
And if the input is not like that then a corresponding converter must be
inserted.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1283>
 2023-07-24 10:14:59 +00:00
Sebastian Dröge
31b1cb8ca6 Update minimum supported Rust version to 1.70 
gtk-rs will update soonish too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1280>
 2023-07-19 09:19:34 +03:00
Mathieu Duponchelle
9707bb89e6 webrtcsink: fix pipeline when input caps contain max-framerate 
GstVideoInfo uses max-framerate to compute its fps, but this leads
to issues in videorate when framerate is actually 0/1.

Fix this by stripping away max-framerate from input caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1276>
 2023-07-13 22:18:08 +02:00
Sebastian Dröge
0331522128 webrtcsink: Configure only 4 threads for x264enc 
More threads can cause more slices to be created, and Chrome simply falls
apart if there are more than a few slices and fails decoding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1275>
 2023-07-13 16:59:43 +03:00
Sebastian Dröge
ca51cf2509 webrtcsink: Translate force-keyunit events to force-IDR action signal for NVIDIA encoders 
NVIDIA's v4l2 encoder elements don't handle the force-keyunit events but
instead provide a custom action signal based API for requesting a
keyframe.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1274>
 2023-07-12 10:09:32 +00:00
Sebastian Dröge
bbd3d9ffe0 Remove unnecessary mut everywhere 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1273>
 2023-07-11 10:09:35 +03:00
Sebastian Dröge
ee4aca3010 webrtcsink: Set config-interval=-1 and aggregate-mode=zero-latency on rtph26[45]pay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
 2023-07-10 19:48:37 +03:00
Sebastian Dröge
957a28f239 webrtcsink: Set VP8/VP9 payloader based on payloader element factory name 
Instead of checking the encoder's name. There are more VP8/VP9 encoders
than the ones from the vpx plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
 2023-07-10 19:45:17 +03:00
Mathieu Duponchelle
1dd13c4812 webrtcsink: fix session_id / peer_id confusion 
In a few places, for instance parameter names, peer_id was still used
when session_id was actually getting passed.

Go through all instances of peer_id in webrtcsink/imp.rs and address
those mix-ups.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1269>
 2023-07-07 05:33:30 +00:00
Bilal Elmoussaoui
dd2d7d9215 Use re-exported once_cell 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 17:50:49 +03:00
Sebastian Dröge
58adebb325 Fix a couple of typos 2023-07-06 13:50:17 +03:00
Olivier Crête
08b6251a7a webrtc-utils: Ensure there is only one cancellable call at a time 
Since we only have one canceller at a time, panic if one try to
use it twice in parallel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Olivier Crête
817b60a758 webrtc: Value.get() is already type checks in the property calls 
GObject will have ensured we get a GValue of the right type.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Olivier Crête
793ee66afa webrtcsink: Add LiveKit WebRTC sink and signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Sebastian Dröge
c350f3c2af webrtcink: Use correct property types for nvvideoconvert 
These are enums and not plain integers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1256>
 2023-06-26 14:48:58 +00:00
Mathieu Duponchelle
84a33ca7b9 webrtcsink: bring in signalling code from whipsink as a signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
 2023-06-16 00:32:56 +02:00
Mathieu Duponchelle
f00a169081 webrtcsrc: add twcc extension to codec-preferences when present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1245>
 2023-06-15 20:41:53 +00:00
Mathieu Duponchelle
1200ae0ee6 webrtcsink: improve debug 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Mathieu Duponchelle
64056c5527 net/webrtc: improve documentation layout 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Sebastian Dröge
8a7a1f519c webrtc: Update to fastrand 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1240>
 2023-06-09 09:36:51 +03:00
Mathieu Duponchelle
81ae675f2d webrtcsink: don't try to use cudaconvert if not present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
 2023-06-08 15:32:49 +02:00
Mathieu Duponchelle
7f78a8428e webrtcsink: dump discovery pipelines on state changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
 2023-06-08 15:32:49 +02:00