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Author SHA1 Message Date
François Laignel
b68e2a1ed0 webrtcsink: remove unneeded mut 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1221>
 2023-05-24 10:18:43 +02:00
Thibault Saunier
04e35e86d6 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/1214>
 2023-05-18 18:23:56 +03:00
Thibault Saunier
e73d7082a6 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/1214>
 2023-05-18 18:23:56 +03:00
François Laignel
7ba0073052 use Pad builders for optional name definition 
Also, apply auto-naming in the following cases

* When building from a non wildcard-named template, the name of the template is
  automatically assigned to the Pad. User can override with a specific name by
  calling `name()` on the `PadBuilder`.
* When building with a target and no name was provided via the above, the
  GhostPad is named after the target.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/448
Auto-naming discussion: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1255#note_1891181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1197>
 2023-05-12 12:55:31 +02:00
François Laignel
680d5221db net/webrtc: src: add signal "request-encoded-filter" 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
 2023-05-09 12:02:15 +02:00
François Laignel
092ae1fec8 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/1202>
 2023-05-09 11:17:32 +02:00
François Laignel
dc5ddd3022 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/1202>
 2023-05-09 10:26:11 +02:00
François Laignel
eca269cbf2 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/1202>
 2023-05-09 10:26:07 +02:00
Sebastian Dröge
cb5b527d74 Update to AWS SDK 0.27 and async-tungstenite 0.22 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1199>
 2023-05-02 15:30:00 +03:00
Sebastian Dröge
5451035215 Update async-tungstenite and AWS SDK dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1187>
 2023-04-21 10:48:10 +00:00
Sebastian Dröge
cc3646640e Fix a couple of new Rust 1.69 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1186>
 2023-04-20 16:47:45 +03:00
Mathieu Duponchelle
dbdb9bc164 webrtcsink: fix navigation data channel 
At some point, presumably recently, the data channel stopped being
requested in Ready, making webrtcbin refuse to create it.

There was quite a lot of churn recently so I couldn't pinpoint the
breaking commit easily.

Fix by simply restoring the correct behavior of requesting the channel
after going to the Ready state

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/341

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1180>
 2023-04-14 14:26:22 +02:00
Mathieu Duponchelle
f1fd8d84c3 webrtc: extract a BaseWebRTCSink 
For documentation purposes, AwsKVSWebRTCSink should not inherit from
another element.

+ Mark base class as plugin API and update plugin cache

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
 2023-04-13 15:06:59 +00:00
Loïc Le Page
dba91bceca webrtc: fix documentation after signaller interface changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1175>
 2023-04-12 20:19:22 +02:00
Thibault Saunier
8f2273328b webrtcsrc: Return bool en 'end-session' as required 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1172>
 2023-04-12 12:17:56 +00:00
Guillaume Desmottes
403004a85e fix typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
 2023-04-10 13:35:32 +02:00
Mathieu Duponchelle
a455819871 webrtcsink: fix tracking of signaller state 
For the signaller to get stopped, we need to remember that we started it
in the first place.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
3368f55a88 webrtcsink: don't return value from error closure 
the signal doesn't expect a return value, which meant we were panicking
as soon as the signaller tried to report an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
58c8c0edc7 webrtc: signaller iface: fix session-ended vs end-session confusion 
Session ending is bidirectional: the signaller can tell the sink that a
session was ended, and the sink can tell the signaller to end a session.

As such, two signals are needed, before this patch the second case was
not working as in essence the sink was telling itself that a session was
ended, and obviously failing to even find it when trying to end it again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Tim-Philipp Müller
7c30430320 webrtc-api: replace LICENSE file symlink with copy 
As in !1157

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1169>
 2023-04-08 17:22:37 +01:00
Matthew Waters
e69b4b7f45 webrtc/signaller/iface: give variables appropriate names 
Rather than arg0, arg1, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
4f4e5f0d75 webrtcsink/signaller: don't call signals while having state/settings locked 
It is a recipe for deadlocks if the signal callback calls back into
webrtcsink in some way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
1c61e46f37 webrtcsink: privatise signalling functions 
The functionality is now access through the relevant signals instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
2ac560975c webrtc/signaller: emit the relevant signals instead of the interface vtable 
In order to support the use case of an external user providing their own
signalling mechanism, we want the signals to be used and only if nothing
is connected, fallback to the default handling.  Calling the interface
vtable directly will bypass the signal emission entirely.

Also ensure that the signals are defined properly for this case. i.e.
1. Signals the the application/external code is expected to emit are
   marked as an action signal.
2. Add accumulators to avoid calling the default class handler if
   another signal handler is connected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
343b659755 webrtc/signaller: remove SignallableImplExt 
This pattern is used for subclassing and calling parent class/interface functions.
However that is not useful for the signaller object.
1. The signals are the API contract and should instead be used by
   webrtcsrc/sink to ask or provide outside for/with information.
2. The default case (no signal attached)is instead handled by default class
   handlers that call directly using the relevant rust trait.  No parent
   (GObject) vfuncs necessary.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
b6e78b5f04 webrtcsink: expose signaller as a property 
in the process move the signaller field to the settings struct

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Thibault Saunier
8236f3e5e7 webrtcsink: Port to the 'webrtcsrc' signaller object/interface 
With contributions from:
Matthew Waters <matthew@centricular.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:03:47 +10:00
Loïc Le Page
f17622a1e1 webrtc: Add gstwebrtc-api subproject in net/webrtc plugin 
This subproject adds a high-level web API compatible with GStreamer
webrtcsrc and webrtcsink elements and the corresponding signaling
server. It allows a perfect bidirectional communication between HTML5
WebRTC API and native GStreamer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/946>
 2023-04-04 16:29:44 +02:00
Tim-Philipp Müller
8845f6a4c6 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/1157>
 2023-04-04 14:26:37 +01:00
Mathieu Duponchelle
15e1844956 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/1151>
 2023-03-31 12:19:07 +00:00
Sebastian Dröge
315e53f064 webrtc: Update to AWS SDK 0.55/0.25 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1152>
 2023-03-31 09:12:26 +00:00
David Revay
002a70a2a4 chore(webrtcsink): fix max-bitrate blurb and nick 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1150>
 2023-03-28 16:11:05 +11:00
Vivia Nikolaidou
7a1b2d97d4 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/1143>
 2023-03-27 16:12:13 +03:00
Sebastian Dröge
c1bac30694 webrtc: Update to aws 0.54/0.24 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1131>
 2023-03-11 09:37:14 +02:00
Mathieu Duponchelle
584392049c net/webrtc: implement AWS KVS signaller 
And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink.

This adds support in webrtcsink for processing a consumer offer, instead
of producing one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
 2023-03-09 15:39:09 +00:00
Sebastian Dröge
fc5ed15af5 Update for gst::Element::link_many() and related API generalization 
Specifically, get rid of now unneeded `&`.
 2023-03-09 16:46:52 +02:00
Thibault Saunier
ce3bb2f1d4 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/932>
 2023-02-28 20:50:15 -03:00
Thibault Saunier
0ae637f531 webrtcsink: Move RUNTIME to the crate so it can be reused 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
 2023-02-28 17:57:14 -03:00
Thibault Saunier
4ec441560b webrtc: Enhance debug messages when using unknown peer ID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
 2023-02-28 19:28:51 +00:00
Matthew Waters
542c7e12b8 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/1107>
 2023-02-28 10:12:36 +11:00
Sebastian Dröge
9fc1404415 Update minimum supported Rust version to 1.66 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
 2023-02-20 11:09:01 +02:00
Sebastian Dröge
ac8afc4ac0 Update to async-tungstenite 0.20 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1087>
 2023-02-10 13:03:07 +02:00
Sebastian Dröge
1e13dbb99c Update versions to 0.11.0-alpha.1 2023-02-10 00:23:56 +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