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Author SHA1 Message Date
Mathieu Duponchelle
927c3e9bdf webrtcsink: don't try to use cudaconvert if not present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:48:08 +03:00
Mathieu Duponchelle
dbd8946608 webrtcsrc: add twcc extension to codec-preferences when present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:46:23 +03:00
Sebastian Dröge
aa799bc26c webrtc: Update to fastrand 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1248>
 2023-06-19 18:43:36 +03:00
Sebastian Dröge
361152d884 Update versions to 0.10.8 2023-06-07 00:54:32 +03:00
Mathieu Duponchelle
da51c3a58b 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/1233>
 2023-06-06 22:38:19 +03:00
Mathieu Duponchelle
2bb0a666a8 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/1233>
 2023-06-06 22:36:51 +03:00
Mathieu Duponchelle
77f003f699 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/1233>
 2023-06-06 22:36:44 +03:00
François Laignel
b8b718fe62 webrtcsink: remove unneeded mut 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233>
 2023-06-06 22:34:55 +03:00
Thibault Saunier
c1d6094bc4 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/1215>
 2023-05-18 18:25:44 +03:00
Thibault Saunier
0e447a9316 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/1215>
 2023-05-18 18:25:44 +03:00
Sebastian Dröge
573307b32e Update version to 0.10.7 2023-05-09 20:44:27 +03:00
Sebastian Dröge
41ea793fd8 Update to AWS SDK 0.27 and async-tungstenite 0.22 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1203>
 2023-05-09 16:00:00 +03:00
François Laignel
91fe56468a net/webrtc: src: add signal "request-encoded-filter" 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176>
 2023-05-02 15:22:43 +02:00
François Laignel
2b6a908911 net/webrtc: sink: add signal "request-encoded-filter" 
The new "request-encoded-filter" signal is emitted when the encoder and related
elements are added to the pipeline. When defined, the element returned by the
signal is inserted between the encoder and the payloader.

The transformation can be reverted using the [insertable streams API] on the
receiver side.

[insertable streams API]: https://developer.mozilla.org/en-US/docs/Web/API/Insertable_Streams_for_MediaStreamTrack_API

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176>
 2023-05-02 15:22:43 +02:00
François Laignel
0e6b9df932 net/webrtc: sink: abort stats collection before stopping the Signaller 
In some rare cases, the webrtc-test entered a deadlock while executing
`WebRTCSink::unprepare`. Attaching gdb to a blocked instance showed:

* `gstrswebrtc::signaller:👿:Signaller::stop()` parked, waiting for a
  `Condvar` in `Signaller::stop()`. This was most likely awaiting for the
  receive task to complete while it was locked in `element.end_session()`.
  This code path is triggered from `unprepare` with the `State` `Mutex` locked.
* `webrtcsink:👿:WebRtcSink::process_stats` waiting for a contended `Mutex`,
  which is also the `State` `Mutex`. This prevented completion of the signal
  `gst_webrtc_bin_get_stats`.

This commit aborts the task in charge of periodically collecting stats and
ensures any remaining iteration completes before requesting the Signaller to
stop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176>
 2023-05-02 15:22:43 +02:00
François Laignel
2cb1fd7fc1 net/webrtc: src: don't set stun-server on webrtcbin when our property is None 
... otherwise an error occurs about the stun-server address being an empty
string which doesn't comply with the expected address format.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176>
 2023-04-30 13:04:26 +02:00
Sebastian Dröge
a29769789f Update async-tungstenite and AWS SDK dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
Sebastian Dröge
5c580709ee Fix a couple of new Rust 1.69 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189>
 2023-04-22 12:18:44 +03:00
François Laignel
d2db786136 net/webrtc: backport the serial-sorted WebRtcSink pad request 
This is a partial backport of [#58439204] to get predictable track order.
With this commit, we are sure the `mid`s sequence in the Sdp offer will reflect
the order by which the `webrtcsink` pads were requested.

[#58439204]: 584392049c
 2023-04-20 15:09:43 +02:00
Sebastian Dröge
48ffd4eb49 Update versions to 0.10.6 2023-04-06 11:24:25 +03:00
Tim-Philipp Müller
198477e63b 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/1160>
 2023-04-05 15:12:40 +00:00
Mathieu Duponchelle
525e3afe70 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/1160>
 2023-04-05 15:12:40 +00:00
David Revay
5bb65d3e33 chore(webrtcsink): fix max-bitrate blurb and nick 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:39 +00:00
Vivia Nikolaidou
f15fd82f83 webrtcsink: Add ice-transport-policy option 
Can be used to force relay ICE candidates, ensuring TURN server is used.
Proxy to the corresponding setting in webrtcbin,

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160>
 2023-04-05 15:12:39 +00:00
Sebastian Dröge
67de191be2 Update versions to 0.10.5 2023-03-19 18:40:49 +02:00
Sebastian Dröge
751a4c7597 Update versions to 0.10.4 2023-03-14 14:01:04 +02:00
Thibault Saunier
ffd70356c1 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/1118>
 2023-03-02 14:14:59 +02:00
Sebastian Dröge
20b18c23e6 Update versions to 0.10.3 2023-03-02 13:27:22 +02:00
Thibault Saunier
528f46a510 webrtcsink: Move RUNTIME to the crate so it can be reused 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1115>
 2023-03-02 11:00:23 +02:00
Thibault Saunier
8b8f10691c webrtc: Enhance debug messages when using unknown peer ID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1115>
 2023-03-02 11:00:23 +02:00
Matthew Waters
84bae8b9cf 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/1115>
 2023-03-02 10:50:19 +02:00
Sebastian Dröge
f22b3420b6 Update versions to 0.10.2 2023-02-23 10:07:43 +02:00
Sebastian Dröge
73a5703eeb Update versions to 0.10.1 2023-02-13 11:52:37 +02:00
Sebastian Dröge
a4d5a35403 Update to async-tungstenite 0.20 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1091>
 2023-02-11 19:44:01 +02:00
Sebastian Dröge
f3c1a91fb9 Add versions to local dependencies 2023-02-10 00:36:22 +02:00
Sebastian Dröge
6df679d69f Update to gtk-rs-core 0.17, gtk4-rs 0.6 and gstreamer-rs 0.20 branches 2023-02-10 00:33:25 +02:00
Sebastian Dröge
85bf8d6c63 Update versions to 0.10.0 2023-02-10 00:26:24 +02:00
Sebastian Dröge
560bdc4cb7 Update for glib API changes 2023-01-31 12:24:07 +02:00
Sebastian Dröge
3b4c48d9f5 Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1062>
 2023-01-25 10:31:19 +02:00
Sebastian Dröge
2c386fb792 Update for various deprecated APIs 2023-01-22 20:07:26 +02:00
Sebastian Dröge
4582ae91ab Move remaining plugins to ParamSpec builders 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1054>
 2023-01-21 18:34:55 +02:00
Sebastian Dröge
812df78b75 webrtcbin: Update for StreamProducer API changes 2023-01-16 16:36:41 +02:00
Sebastian Dröge
6132788b02 Update for caps/structure-related string API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1048>
 2023-01-15 22:58:44 +02:00
Mathieu Duponchelle
1a8abde884 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/1044>
 2023-01-11 15:09:45 +00:00
Sebastian Dröge
27435ad82e Update for API changes 2023-01-05 12:33:54 +02:00
Zhao, Gang
9fa838e366 webrtc: Fix rustfmt errors 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-27 11:12:54 +02:00
Zhao, Gang
877a9bd7f3 webrtc: Share runtime between webrtcsink and signaller crates 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00
Zhao, Gang
1ffeb4d44d webrtc: Move from async-std to tokio 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00
Zhao, Gang
2bc29c1fd3 webrtc: examples: Update package-lock.json 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00
Mathieu Duponchelle
e5360ff431 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/1012>
 2022-12-13 12:47:26 +01:00