mirrors/gst-plugins-rs
mirrors/gst-plugins-rs
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git synced 2024-11-15 23:01:02 +00:00
Code Issues Projects Releases Wiki Activity
2547 commits 14 branches 100 tags 38 MiB
Commit graph

52 commits

Author SHA1 Message Date
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
Thibault Saunier
cbdd3a7f26 webrtc: Enhance documentation 2022-10-20 12:04:43 +00:00
Sebastian Dröge
c0bf05d4bb webrtc: Minor cleanup 2022-10-20 13:20:32 +03:00
Thibault Saunier
71ed04d89b webrtc: Rename signaller and protocol crates 2022-10-20 13:32:31 +02:00
Thibault Saunier
25bda89ac8 webrtc: Update an unify rust-version and edition 
So it all matches the rest of the plugins
 2022-10-20 13:32:31 +02:00
Thibault Saunier
4942a916a8 webrtc: Uniformise GType names 2022-10-20 13:32:31 +02:00
Thibault Saunier
37c0239aff webrtc: Port to new ElementBuilder API 2022-10-20 13:32:31 +02:00
Thibault Saunier
ad78936365 webrtc: Enable more documentation 2022-10-20 13:32:31 +02:00
Thibault Saunier
0f0dec7fa9 webrtc: Fix fmt issues 2022-10-20 11:51:59 +02:00
Thibault Saunier
5ab7be6124 webrtc: Add SDPX license header on every file 2022-10-20 11:51:58 +02:00
Thibault Saunier
39c0dcb0d4 Plug webrtc in 2022-10-20 11:51:58 +02:00
Thibault Saunier
b164daf510 webrtc: Fix clippy issues 2022-10-20 11:51:58 +02:00