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496 commits

Author SHA1 Message Date
Piotr Brzeziński e792fdbe48 webrtc: Fix paths in README 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
 2023-10-04 19:07:14 +03:00
Sean DuBois 4a1a7864cf net: webrtc/webrtchttp: Respect HTTP redirects 
Properly follow redirect URL. Before new request would be made, but with
original URL again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
 2023-10-04 19:06:03 +03:00
Sebastian Dröge a168456428 Update to AWS SDK 0.31 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
 2023-10-04 19:04:21 +03:00
Sebastian Dröge 9187d0e11e onvifmetadataparse: Skip metadata frames with unrepresentable UTC time 
Previously we would panic, which causes the element to post an error
message. Instead, simply skip metadata frames if their UTC time since
the UNIX epoch can't be represented as nanoseconds in u64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1331>
 2023-09-20 19:58:43 +03:00
Seungha Yang 468f73092d 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/1331>
 2023-09-20 19:58:13 +03:00
Seungha Yang 2b1d07a757 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/1331>
 2023-09-20 19:58:06 +03:00
Mathieu Duponchelle 33156b52c8 webrtcsink: fix codec selection discoveries 
Since ab1ec126983f949804684e11e0e58c7cf3b22bc4:

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/1331>
 2023-09-20 19:57:57 +03:00
Robert Ayrapetyan 391ba3f5f8 webrtcsink: fix TWCC extension adding 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1331>
 2023-09-20 19:57:40 +03:00
Sebastian Dröge dc1e45eaa1 Update to AWS SDK 0.30 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1331>
 2023-09-20 19:57:00 +03:00
Sebastian Dröge f35603a020 webrtc: Add versions to gst_plugin_webrtc_protocol dependency 2023-08-10 16:57:11 +03:00
Sebastian Dröge 982b5d6ef9 Add version to the gst-plugin-version-helper dependency 2023-08-10 16:56:04 +03:00
Sebastian Dröge 67c3732b94 Update dependencies to release branches of gtk-rs / gtk4-rs / gstreamer-rs 2023-08-09 18:14:18 +03:00
Sebastian Dröge 9b1853c1de Update versions to 0.11.0 2023-08-09 18:04:38 +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
Sebastian Dröge 2591feb72e Update a couple of dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1294>
 2023-08-07 11:42:32 +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
Bilal Elmoussaoui 2cc98bf410 Adapt to glib::Continue rename 
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
Seungha Yang 1f0ce101eb awstranscriber: Tone down log message 
It's not an ERROR case at all

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1263>
 2023-06-28 23:57:54 +09:00
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
Mathieu Duponchelle 7447d95f1b webrtc/signalling: fix race condition in message ordering 
Spawning one task per message to send out instead of sending them out
sequentially from the one task used to poll the handler sometimes
resulted in peers receiving ICE candidates before SDP offers, triggering
hard to understand errors in the browser.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 13:24:45 +02:00
Mathieu Duponchelle de0f7a08fe gstwebrtc-api: fix firefox errors about more than two stun servers 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 13:24:45 +02:00
Mathieu Duponchelle cd4b90fef4 webrtcsink/utils: remove unused decoders field in DecodingInfo 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 01:54:13 +02:00
Mathieu Duponchelle 271b583876 webrtcsink: avoid panic on unprepare from an async tokio context 
.. and log an error with advice on how to dispose of elements properly
from a tokio runtime.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1218>
 2023-06-07 19:57:19 +00:00
Sebastian Dröge c65b3429ad Use MPL as license specifier for plugins only requiring GStreamer < 1.20 
And use MPL-2.0 for all that require GStreamer 1.20 or newer. The new
string is only allowed in 1.20 or newer and using it in older versions
causes failure to load the plugin.

All affected plugins are of course still MPL-2.0 licensed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/374

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1235>
 2023-06-07 19:13:55 +03:00