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
bea00c7413
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/1248 >
2023-06-19 18:42:12 +03:00
Sebastian Dröge
361152d884
Update versions to 0.10.8
2023-06-07 00:54:32 +03:00
Mathieu Duponchelle
1edf4a144e
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/1233 >
2023-06-06 22:16:55 +02:00
Edward Hervey
18773a9df1
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/1233 >
2023-06-06 22:43:33 +03:00
Sebastian Dröge
e8e247d1ed
net: Update to AWS SDK 0.28
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233 >
2023-06-06 22:41:20 +03:00
Sebastian Dröge
70f92ddbf7
whipsink: Request pads with webrtcbin's pad templates and not our own
...
It's invalid to request pads with a pad template that is not part of the
element, and results in a critical warning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1233 >
2023-06-06 22:40:01 +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
1db07fe451
aws: Update to AWS SDK 0.55/0.25
...
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
Edward Hervey
a76330e76c
rtpgccbwe: Don't process empty lists
...
The structure parsing could result in an empty vector. Don't do any processing
since the loss code assumes it's non-empty for average estimates which would
result in weird/invalid results.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189 >
2023-04-22 11:56:37 +03:00
Sebastian Dröge
33be56bd26
net: ndi: Update to libloading 0.8
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1189 >
2023-04-22 11:56:13 +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
Sebastian Dröge
0d8a5245b0
ndisrc: Fix copying of raw video frames with different NDI/GStreamer strides
...
And also don't copy each line twice for single-plane formats.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1160 >
2023-04-05 15:12:40 +00:00
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
Sebastian Dröge
3a394e0118
Fix a few new clippy warnings
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1129 >
2023-03-10 16:48:32 +01:00
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
Vivia Nikolaidou
3dde725560
ndisinkcombiner: Properly handle caps changes
...
We are caching one video buffer, so previously we were changing the src
caps one buffer too early.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1115 >
2023-03-02 11:00:23 +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
Arun Raghavan
54adcb8482
hlssink3: Allow GIOStream signal handlers to return None
...
If creating a playlist or fragment stream fails (disk is full, the
directory is removed, ...), we will currently crash because the signal
handler expects a non-None GIOStream. The actual callback is allowed to
return None values and we handle this in the caller, so let's not have
this restriction on the signal handler.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1097 >
2023-02-21 16:16:16 +02:00
Sebastian Dröge
73a5703eeb
Update versions to 0.10.1
2023-02-13 11:52:37 +02:00
Seungha Yang
a505dba3a2
rtpav1pay: Fix Leb128Bytes size parsing
...
There are multiple ways of encoding the value, and don't assume
that bitstream used the way used in this plugin. Instead, count
the number of used bytes.
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/312
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1091 >
2023-02-11 19:44:08 +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