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Author SHA1 Message Date
Robert Ayrapetyan
e83238b681 webrtcsink: fix TWCC extension adding 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1310>
 2023-09-04 18:27:51 +00:00
Sebastian Dröge
8d433761d1 Fix indentation of let-else blocks 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1308>
 2023-08-29 12:21:38 +00:00
Taruntej Kanakamalla
de6d2e7f40 net/webrtc: rename whipwebrtcsink as whipclientsink 
add a deprecation warning in whipsink to indicate it
should be used only for RTP content
add documentation in whipsink code regarding usage and
deprecation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1282>
 2023-08-26 10:53:30 +05:30
Sebastian Dröge
905da44958 Update to AWS SDK 0.30 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1313>
 2023-08-25 09:46:52 +03:00
Andoni Morales Alastruey
3c1f05cdc3 webrtcsrc: document how to use the element for remote control 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
 2023-08-10 17:43:51 +00:00
Andoni Morales Alastruey
3000b08ec7 webrtcsrc: add support for navigation events 
This provides support GstNavigation events handling in webrtcsrc so that
a GStreamer client can be used to control remotely a GStreamer server,
similar to how the web client is capable of controlling a wpesrc.
This is part of a larger set of patches that require more work on the
sinks and sources.
server: d3d11screencapturesrc ! webrtcsink enable-data-channel-navigation=true
client: webrtcsrc enable-data-channel-navigation=true ! d3d11videosink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
 2023-08-10 17:43:51 +00:00
Loïc Le Page
e5e3dc6e19 net/webrtc/signaller: add property to get the connection client ID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
 2023-08-10 17:30:21 +02:00
Loïc Le Page
7af2ff0843 net/webrtc/signaller: advertise running producers in Listener mode 
When starting a webrtcsrc-signaller client in Listener mode, only the producers
started after the client connection were advertised. All currently
running producers were ignored unlike the gstwebrtc-api behavior. This
commit now lists all running producers when the client Listener connects
and advertises them through the "producer-added" signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
 2023-08-10 17:30:21 +02:00
Sebastian Dröge
d688aeb184 Update versions to 0.12.0-alpha.1 2023-08-10 17:21:11 +03:00
Sebastian Dröge
3b41f206bc Don't generate .def files for plugins 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
 2023-08-09 13:54:34 +03:00
Sebastian Dröge
b3826c108d webrtc: Update to async-tungstenite 0.23 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
 2023-08-09 13:18:44 +03:00
Sebastian Dröge
5ee46a214c webrtc: Use #[repr(C)] to get a C-compatible layout for the Signaller struct 
This is required by GObject for class/interface and instance structs and
the reason why implementing the `glib::ObjectInterface` trait is unsafe.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/397

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1297>
 2023-08-09 10:32:44 +03:00
Sebastian Dröge
cac791a6ca aws/webrtc: Update to AWS SDK 0.56/0.29 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1295>
 2023-08-07 20:03:51 +03:00
Sanchayan Maity
5b60ecbb18 net: webrtc/webrtchttp: Fix canceller usage 
Commit 08b6251a added the check to ensure only one canceller at a time for net/webrtc.

In `whipsink` and since `whipwebrtcsink` picked up the same implementation, there exists a
bug around the use of canceller. `whipsink` calls `wait_async` while passing the canceller
as an argument. The path `send_offer -> do_post -> parse_endpoint_response` results in the
canceller being replaced in each subsequent call to `wait_async`. Since `wait_async` call
does not ensure one canceller, with the async call the use of canceller/abort was subtly
broken. Similarly, for `whepsrc`.

We really don't need to use `wait_async` inside `do_post` for any `await` calls. If the
root future viz. `do_post` with `wait_async` is aborted, the child futures will be taken
care of.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1290>
 2023-08-04 10:01:11 +05:30
Mathieu Duponchelle
9680805bdb 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/1289>
 2023-08-01 00:28:52 +02:00
Mathieu Duponchelle
dbeb65da06 webrtc/utils: fix typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1289>
 2023-08-01 00:28:32 +02:00
Sebastian Dröge
d4b3827efa 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/1283>
 2023-07-24 10:14:59 +00:00
Sebastian Dröge
31b1cb8ca6 Update minimum supported Rust version to 1.70 
gtk-rs will update soonish too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1280>
 2023-07-19 09:19:34 +03:00
Mathieu Duponchelle
9707bb89e6 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/1276>
 2023-07-13 22:18:08 +02:00
Sebastian Dröge
0331522128 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/1275>
 2023-07-13 16:59:43 +03:00
Sebastian Dröge
ca51cf2509 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/1274>
 2023-07-12 10:09:32 +00:00
Sebastian Dröge
bbd3d9ffe0 Remove unnecessary mut everywhere 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1273>
 2023-07-11 10:09:35 +03:00
Sebastian Dröge
ee4aca3010 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/1272>
 2023-07-10 19:48:37 +03:00
Sebastian Dröge
957a28f239 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/1272>
 2023-07-10 19:45:17 +03:00
Mathieu Duponchelle
1dd13c4812 webrtcsink: fix session_id / peer_id confusion 
In a few places, for instance parameter names, peer_id was still used
when session_id was actually getting passed.

Go through all instances of peer_id in webrtcsink/imp.rs and address
those mix-ups.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1269>
 2023-07-07 05:33:30 +00:00
Bilal Elmoussaoui
dd2d7d9215 Use re-exported once_cell 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 17:50:49 +03:00
Sebastian Dröge
58adebb325 Fix a couple of typos 2023-07-06 13:50:17 +03:00
Olivier Crête
08b6251a7a webrtc-utils: Ensure there is only one cancellable call at a time 
Since we only have one canceller at a time, panic if one try to
use it twice in parallel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Olivier Crête
817b60a758 webrtc: Value.get() is already type checks in the property calls 
GObject will have ensured we get a GValue of the right type.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Olivier Crête
793ee66afa webrtcsink: Add LiveKit WebRTC sink and signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Sebastian Dröge
c350f3c2af 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/1256>
 2023-06-26 14:48:58 +00:00
Mathieu Duponchelle
84a33ca7b9 webrtcsink: bring in signalling code from whipsink as a signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
 2023-06-16 00:32:56 +02:00
Mathieu Duponchelle
f00a169081 webrtcsrc: add twcc extension to codec-preferences when present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1245>
 2023-06-15 20:41:53 +00:00
Mathieu Duponchelle
1200ae0ee6 webrtcsink: improve debug 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Mathieu Duponchelle
64056c5527 net/webrtc: improve documentation layout 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Sebastian Dröge
8a7a1f519c webrtc: Update to fastrand 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1240>
 2023-06-09 09:36:51 +03:00
Mathieu Duponchelle
81ae675f2d webrtcsink: don't try to use cudaconvert if not present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
 2023-06-08 15:32:49 +02:00
Mathieu Duponchelle
7f78a8428e webrtcsink: dump discovery pipelines on state changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
 2023-06-08 15:32:49 +02:00
Mathieu Duponchelle
7447d95f1b 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/1236>
 2023-06-08 13:24:45 +02:00
Mathieu Duponchelle
de0f7a08fe gstwebrtc-api: fix firefox errors about more than two stun servers 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 13:24:45 +02:00
Mathieu Duponchelle
cd4b90fef4 webrtcsink/utils: remove unused decoders field in DecodingInfo 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 01:54:13 +02:00
Mathieu Duponchelle
271b583876 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/1218>
 2023-06-07 19:57:19 +00:00
Mathieu Duponchelle
fda5aed89f webrtcsink: encoded streams: address last review comments 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 16:05:28 +02:00
Thibault Saunier
ab1ec12698 webrtcsink: Add support for pre encoded streams 
This is a first step where we try to replicate encoding conditions from
the input stream into the discovery pipeline. A second patch will
implement using input buffers in the discovery pipelines.

This moves discovery to using input buffers directly. Instead of trying
to replicate buffers that `webrtcsink` is getting as input with testsrc,
directly run discovery based on the real buffers. This way we are sure
we work with the exact right stream type and we don't need encoders to
support encoding streams inputs.

We use the same logic for both encoded and raw input to avoid having
several code paths and makes it all more correct in any case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:32:40 +02:00
Thibault Saunier
059cdecf7d webrtc: Unify the Codec structure between sink and source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:31:45 +02:00
Thibault Saunier
cf32d9d668 webrtc: Move make_element to the utils 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:31:45 +02:00
Thibault Saunier
ce42723ad2 webrtc: Minor documentation enhancement 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:31:45 +02:00
Mathieu Duponchelle
80582923bb aws_kvs_signaller: don't force us-east-1 region 
Instead use default region provider, with a fallback to us-east-1

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1228>
 2023-05-30 16:04:27 +00:00
François Laignel
4cc2498c24 webrtcsink: use spawn_blocking instead of call_async 
In `webrtcsink`, we terminate a session by setting the session's pipeline to
`Null` like this:

```rust
    pipeline.call_async(|pipeline| {
        [...]
        pipeline.set_state(gst::State::Null);
        [...]
        // the following cvar is awaited in unprepare()
        cvar.notify_one();
    });
```

However, `pipeline.call_async` keeps a ref on the pipeline until it's done,
which means the `cvar` is notified before `pipeline` is actually 'disposed',
which happens in a different thread than `unprepare`'s. [`gst_rtp_bin_dispose`]
releases some resources when the pipeline is unrefed. In some cases, those
resources are actually released after the main thread has returned, leading
various issues.

This commit uses tokio runtime's `spawn_blocking` instead, which allows owning
and disposing of the pipeline before the `cvar` is notified.

[`gst_rtp_bin_dispose`]: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/blob/main/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpbin.c#L3108

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1225>
 2023-05-26 14:23:03 +02:00
Mathieu Duponchelle
a20855dfd9 webrtcsink: expose consumer-pipeline-created signal 
This signal is emitted as soon as the pipeline for each consumer
is created, and can be used by applications that require a greater
level of control over webrtcsink's internals.

An example is also provided to demonstrate usage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1220>
 2023-05-25 13:15:52 +02:00
Sebastian Dröge
a27be7d054 net: Update to AWS SDK 0.28 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1224>
 2023-05-25 13:23:49 +03:00
François Laignel
e62e9f5bd4 webrtcsink: adapt commit "abort stats collection before stopping the Signaller" 
Adapt a commit [1] that was introduced as part of the forward port of the MR
'add signal "request-encoded-filter"' [2].

The deadlock said commit was fixing doesn't happen on main branch due to
changes in the element design: the Sessions are no longer aborted with the
element `State` held. However, we want to ensure the stats collection task
is terminated when the `webrtcbin` element returns from the Ready to Null
transition, meaning that the related resources are released.

[1]: gstreamer/gst-plugins-rs!1176 (0e6b9df9)
[2]: gstreamer/gst-plugins-rs!1176

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1222>
 2023-05-24 21:35:39 +02:00
Mathieu Duponchelle
44a395f134 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/1217>
 2023-05-24 13:35:26 +02:00
Mathieu Duponchelle
e13124a426 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/1217>
 2023-05-24 13:35:26 +02:00
Mathieu Duponchelle
ccf076ed1e 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/1217>
 2023-05-24 13:35:26 +02:00
François Laignel
9a59763df1 webrtcsink: wait for Sessions to end 
`State::finalize_session()` asynchronously sets the Session pipeline to Null.
In some cases, sessions `webrtcbin` could terminate their transition to Null
after `webrtcsink` had reached Null.

This commit adds a set of `finalizing_sessions`. When the finalization process
starts, the session is added to the set. After `webrtcbin` has reached the Null
state, the session is removed from the set and a condvar is notified.

In `unprepare`, `webrtcsink` loops until the `finalizing_sessions` set is
empty, awaiting for the condvar to be notified when it's not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1221>
 2023-05-24 10:18:47 +02:00
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
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
Sebastian Dröge
3f904553ea Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1011>
 2022-12-13 11:43:16 +02:00
Sebastian Dröge
fb42cd8a0f net: Update to async-tungstenite 0.19 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1005>
 2022-12-11 12:54:24 +02:00
Raphael Dürscheid
aa2abc50bf 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/983>
 2022-12-02 10:18:27 +00:00
Sebastian Dröge
fceacf7081 Update for gst::Array / gst::List API improvements 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/985>
 2022-11-27 01:12:46 +02:00
Thibault Saunier
6b11284e8a webrtcsink: Make the turn-server prop a turn-servers list 
So that we can simply specify several turn servers at once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/973>
 2022-11-16 14:48:16 +00:00
Guillaume Desmottes
37cb636140 webrtc: README: fix couple of links 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/975>
 2022-11-11 14:51:46 +01:00
Mathieu Duponchelle
66e7b314f7 webrtcsink: improve debug 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/972>
 2022-11-10 15:00:19 +00:00
Sebastian Dröge
a5f3197651 Add missing doc features to WebRTC plugins 2022-11-07 18:06:29 +00:00
Sebastian Dröge
a8250abbf1 Fix various new clippy warnings 2022-11-01 10:27:48 +02:00
Sebastian Dröge
976ae5707e webrtc: Update to human_bytes 0.4 2022-10-31 14:11:29 +02:00
Sebastian Dröge
bf6bdab80c webrtc: Remove version requirement from internal crate dependencies 2022-10-24 19:50:24 +03:00
Sebastian Dröge
f2223cf2cb Update versions to 0.10.0-alpha.1 2022-10-24 19:31:19 +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
Thibault Saunier
87fd49a9bf webrtc:signalling: Remove short option for 'host' in the cli 
It clashes with `--help`
 2022-10-20 11:51:58 +02:00
Thibault Saunier
5e7537953c webrtc: Move to net/webrtc 2022-10-18 15:18:53 +02:00