Commit graph

488 commits

Author SHA1 Message Date
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
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
Mathieu Duponchelle fda5aed89f webrtcsink: encoded streams: address last review comments
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 16:05:28 +02:00
Thibault Saunier ab1ec12698 webrtcsink: Add support for pre encoded streams
This is a first step where we try to replicate encoding conditions from
the input stream into the discovery pipeline. A second patch will
implement using input buffers in the discovery pipelines.

This moves discovery to using input buffers directly. Instead of trying
to replicate buffers that `webrtcsink` is getting as input with testsrc,
directly run discovery based on the real buffers. This way we are sure
we work with the exact right stream type and we don't need encoders to
support encoding streams inputs.

We use the same logic for both encoded and raw input to avoid having
several code paths and makes it all more correct in any case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 15:32:40 +02:00
Thibault Saunier 059cdecf7d webrtc: Unify the Codec structure between sink and source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 15:31:45 +02:00
Thibault Saunier cf32d9d668 webrtc: Move make_element to the utils
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 15:31:45 +02:00
Thibault Saunier ce42723ad2 webrtc: Minor documentation enhancement
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 15:31:45 +02:00
Mathieu Duponchelle 6346d5608e net/aws/transcriber: track discont offset in input stream
and add it up to subsequent transcripts.

This ensures synchronization is maintained even after the input stream
experiences a discontinuity and a gap in its timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1230>
2023-06-02 08:55:11 +00:00
Mathieu Duponchelle 80582923bb aws_kvs_signaller: don't force us-east-1 region
Instead use default region provider, with a fallback to us-east-1

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1228>
2023-05-30 16:04:27 +00:00
Edward Hervey 31b06e52ea rtpgccbwe: Improve packet handling
Both the delay-based *and* loss-based estimates should be computed instead of
just one. This ensures faster adaptation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1179>
2023-05-29 08:20:36 +00:00