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Author SHA1 Message Date
Sebastian Dröge 7c59caf3f8 webrtcsink: Fix clippy warning 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1330>
 2023-09-20 19:34:07 +03:00
Sebastian Dröge 26d90191b5 onvifmetadataparse: Skip metadata frames with unrepresentable UTC time 
Previously we would panic, which causes the element to post an error
message. Instead, simply skip metadata frames if their UTC time since
the UNIX epoch can't be represented as nanoseconds in u64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1330>
 2023-09-20 19:25:31 +03:00
Seungha Yang d134e165c5 webrtcsink: Propagate GstContext messages 
Implement CustomBusStream so that NEED_CONTEXT and HAVE_CONTEXT
messages from session/discovery can be forwarded to parent
pipeline and also GstContext can be shared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1330>
 2023-09-20 19:24:38 +03:00
Seungha Yang 938d3d73b9 webrtcsink: Add support for d3d11 memory and qsvh264enc 
Adding d3d11 memory and qsvh264enc support

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1330>
 2023-09-20 19:24:31 +03:00
Robert Ayrapetyan 1ec74d8569 webrtcsink: fix TWCC extension adding 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1330>
 2023-09-20 19:24:06 +03:00
Mathieu Duponchelle 3eab53be85 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/1330>
 2023-09-20 19:23:09 +03:00
Sebastian Dröge 23e7226c94 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/1330>
 2023-09-20 19:21:27 +03:00
Sebastian Dröge 6393572b37 Update version to 0.10.11 2023-07-20 14:43:13 +03:00
Mathieu Duponchelle 584dff7961 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/1286>
 2023-07-19 09:53:00 +03:00
Sebastian Dröge acff5a9394 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/1286>
 2023-07-19 09:52:55 +03:00
Sebastian Dröge 369e555e13 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/1286>
 2023-07-19 09:52:49 +03:00
Sebastian Dröge 12bb9afd81 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/1286>
 2023-07-19 09:52:43 +03:00
Sebastian Dröge adb113a591 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/1286>
 2023-07-19 09:52:37 +03:00
Sebastian Dröge 4f78e7b92e Update versions to 0.10.10 2023-07-05 15:53:20 +03:00
Sebastian Dröge 698db4b13e Correctly declare 1.64 as minimum supported Rust version 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1266>
 2023-07-05 12:36:57 +03:00
Sebastian Dröge 0c3def8b9e 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/1266>
 2023-07-05 12:36:08 +03:00
Mathieu Duponchelle 12f1f5b097 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/1253>
 2023-06-20 22:27:06 +02:00
Mathieu Duponchelle d3cda3dd3a 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/1253>
 2023-06-20 22:24:28 +02:00
Sebastian Dröge ab8525451a Update versions to 0.10.9 2023-06-19 20:43:14 +03:00
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