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63 commits

Author SHA1 Message Date
Sebastian Dröge
fc75502ee4 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/1285>
 2023-07-19 09:38:44 +03:00
Sebastian Dröge
9854b299a2 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/1285>
 2023-07-19 09:38:28 +03:00
Sebastian Dröge
bdcdbfeaaf 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/1285>
 2023-07-19 09:38:18 +03:00
Sebastian Dröge
482b7e1469 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/1285>
 2023-07-19 09:38:14 +03:00
Sebastian Dröge
fafe52475f 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/1267>
 2023-07-05 14:41:21 +03:00
Mathieu Duponchelle
0954af10c7 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/1254>
 2023-06-20 22:30:01 +02:00
Sebastian Dröge
dfe2442c92 webrtc/signalling: Allow unknown clippy lints 
tracing is adding some that require a newer Rust version than used here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1249>
 2023-06-19 20:37:53 +03:00
Mathieu Duponchelle
82f3910453 webrtcsink: don't try to use cudaconvert if not present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1249>
 2023-06-19 19:03:04 +03:00
Mathieu Duponchelle
55a6609fdb webrtcsrc: add twcc extension to codec-preferences when present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1249>
 2023-06-19 19:02:48 +03:00
Sebastian Dröge
05b2caec74 webrtc: Update to fastrand 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1249>
 2023-06-19 19:02:16 +03:00
Mathieu Duponchelle
8248425905 webrtcsink: further refactor connection to stats signals 
- Stop passing webrtcbin around without using it

- Stop using glib::closure as clippy complains when using a unit type
  default-return

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1234>
 2023-06-06 22:52:43 +03:00
Mathieu Duponchelle
e9d32fb221 webrtcsink: fix stats_sigid logic 
First off, we just created the session, we know stats_sigid is None
at this point.

Second, don't first assign the result of connecting on-new-ssrc to the
field, then the result of connection twcc-stats, that simply doesn't
make sense.

Finally, actually check that stats_sigid *is* None before connecting
twcc-stats, as I understand it this must have been the original
intention / behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1234>
 2023-06-06 22:52:34 +03:00
Mathieu Duponchelle
8ff2c6609c webrtcsink: don't panic in twcc-stats callback 
If webrtcbin was disposed of at this point, simply return

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/345
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1234>
 2023-06-06 22:52:22 +03:00
Thibault Saunier
0b65a2f8af webrtcsrc: Do not pass raw caps in the transceiver 
That was not making sense.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1216>
 2023-05-18 18:30:08 +03:00
Thibault Saunier
482ff879a4 webrtcsrc: Fix caps used when creating transceiver 
We used to pass all media keys and attributes to the caps which
incorrect. Instead we should be using only the keys from the map
and remove all information related to rtcp which is irrelevant
to create the transceiver.

This also simplifies the code.

New caps look like:

```
Caps(
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 96,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP8",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 102,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 104,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 106,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 108,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 127,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 39,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 98,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "0",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 100,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "2",
    },
)
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1216>
 2023-05-18 18:30:08 +03:00
Tim-Philipp Müller
2d56989f5c 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/1161>
 2023-04-05 18:17:16 +03:00
Mathieu Duponchelle
c2d6273786 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/1161>
 2023-04-05 18:16:10 +03:00
Thibault Saunier
4b867d27fe 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/1117>
 2023-03-02 14:56:30 -03:00
Sebastian Dröge
9a779607c7 Update versions to 0.9.10 2023-03-02 13:18:00 +02:00
Thibault Saunier
e4c9ba43df webrtc: Enhance debug messages when using unknown peer ID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1116>
 2023-03-02 11:01:18 +02:00
Matthew Waters
0d3dc25414 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/1116>
 2023-03-02 10:53:19 +02:00
Sebastian Dröge
eb3d3b3088 Update versions to 0.9.9 2023-02-09 22:08:17 +02:00
Sebastian Dröge
5c2582d105 Update version to 0.9.8 2023-01-23 11:30:27 +02:00
Sebastian Dröge
4ba452dcc3 Update versions to 0.9.7 2023-01-19 19:06:43 +02:00
Sebastian Dröge
c818a575b4 Update versions to 0.9.6 2023-01-18 17:19:17 +02:00
Mathieu Duponchelle
53ae335d22 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/1045>
 2023-01-11 18:38:13 +02:00
Sebastian Dröge
2a8a90f76f Update versions to 0.9.5 2023-01-07 16:06:17 +02:00
Sebastian Dröge
b0bd55c4d2 Update versions to 0.9.4 2022-12-27 13:14:59 +02:00
Sebastian Dröge
bae5294e8f Update versions to 0.9.3 2022-12-16 20:22:17 +02:00
Mathieu Duponchelle
fffd7dc542 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/1017>
 2022-12-16 18:51:08 +02:00
Sebastian Dröge
b4185134d1 Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1017>
 2022-12-16 18:51:00 +02:00
Sebastian Dröge
8c27aefe76 net: Update to async-tungstenite 0.19 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:39:38 +02:00
Raphael Dürscheid
184f879bf7 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/1008>
 2022-12-12 13:37:00 +02:00
Sebastian Dröge
1f4a035dc0 Update versions to 0.9.2 2022-11-28 11:44:33 +02:00
Sebastian Dröge
e434fd19ca Update versions to 0.9.1 2022-11-13 20:23:47 +02:00
Guillaume Desmottes
331d053516 webrtc: README: fix couple of links 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/977>
 2022-11-12 15:52:50 +00:00
Mathieu Duponchelle
5c9bc03eab webrtcsink: improve debug 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/977>
 2022-11-12 15:52:50 +00:00
Sebastian Dröge
2e3373647a Add missing doc features to WebRTC plugins 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/977>
 2022-11-12 15:52:50 +00:00
Sebastian Dröge
07f3b0f504 Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/977>
 2022-11-12 15:52:49 +00:00
Sebastian Dröge
f2f0eb30e0 webrtc: Update to human_bytes 0.4 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/977>
 2022-11-12 15:52:49 +00:00
Sebastian Dröge
ba5270d30a Update to release versions of gtk-rs and gstreamer-rs 2022-10-24 19:28:41 +03:00
Sebastian Dröge
b64f951160 Update to async-tungstenite 0.18 2022-10-24 18:03:33 +03:00
Sebastian Dröge
9a68f6e221 Move from imp.instance() to imp.obj() 
It's doing the same thing and is shorter.
 2022-10-23 23:08:46 +03:00
François Laignel
86776be58c Remove & for obj in log macros 
This is no longer necessary.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1137
 2022-10-23 21:22:31 +02:00
Sebastian Dröge
f045099fc1 Fix GObject type names, GStreamer debug category names and element factory names 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/198
 2022-10-23 20:46:08 +03:00
Sebastian Dröge
5d44e0eb3c rtp: Move GCC bandwidth estimation element from webrtc to rtp plugin 2022-10-23 20:25:08 +03:00
Sebastian Dröge
20ad9175d8 Make GStreamer plugin/crate/library/directory names and descriptions consistent 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/238
 2022-10-23 20:25:08 +03:00
Sebastian Dröge
f058a5e229 Various minor cleanups 2022-10-22 19:50:24 +03:00
François Laignel
6319d104a8 Take advantage of Into<Option<_>> args 
Commit 24b7cfc8 applied changes related to nullability as declared
by gir. One consequence was that some functions signature ended up
requiring users to pass `Some(val)` when they could use `val`
before.

This commit applies changes on `gstreamer-rs` which, will honoring
the nullability stil allow users to pass `val` for the few affected
functions.

This commit also fixes the signature for `Element::request_new_pad`
which was updated upstream.
 2022-10-21 11:54:24 +02:00
Thibault Saunier
5c89c3db69 webrtc: Rename and add to meson build the signalling server 
The binary was only called `server` it has been renamed to
`gst-webrtc-signalling-server` and is installed in meson.
 2022-10-20 18:20:15 +00:00