Commit graph

3174 commits

Author SHA1 Message Date
Robert Ayrapetyan
972b9e5474 doc: add docstrings for signaller object
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
Robert Ayrapetyan
7a72b2fc25 webrtcsink-signalling: add headers support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
François Laignel
91bfd0f7c3 webrtc: signallers: attempt to close the ws when an error occurs
This commit discards the early error returns in the send tasks to log the error
and attempt to close the websocket.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1435>
2024-02-01 18:08:41 +01:00
François Laignel
f54d714afd webrtc: only use close() to close websockets
In the signaller clients and servers, the following sequence is used to close
the websocket (in the [send task]):

```rust
    ws_sink.send(WsMessage::Close(None)).await?;
    ws_sink.close().await?;
```

tungstenite's [`WebSocket::close()` doc] states:

> Calling this function is the same as calling `write(Message::Close(..))``

So we might think they are redundant and either could be used for this purpose
(`send()` calls `write()`, then `flush()`).

The result is actually is bit different as `write()` starts by checking the
state of the connection and [returns `SendAfterClosing`] if the socket is no
longer active, which is the case when a closing request has been received from
the peer via a [call to `do_close()`]). Note that `do_close()` also enqueues a
`Close` frame.

This behaviour is visible from the server's logs:

```
1. tungstenite::protocol: Received close frame: None
2. tungstenite::protocol: Replying to close with Frame { header: FrameHeader { .., opcode: Control(Close), .. }, payload: [] }
3. gst_plugin_webrtc_signalling::server: Received message Ok(Close(None))
4. gst_plugin_webrtc_signalling::server: connection closed: None this_id=cb13892f-b4d5-4d59-95e2-b3873a7bd319
5. remove_peer{peer_id="cb13892f-b4d5-4d59-95e2-b3873a7bd319"}: gst_plugin_webrtc_signalling::server: close time.busy=285µs time.idle=55.5µs
6. async_tungstenite: websocket start_send error: WebSocket protocol error: Sending after closing is not allowed
```

1: The server's websocket receives the peer's `Close(None)`.
2: `do_close()` enqueues a `Close` frame.
3: The incoming `Close(None)` is handled by the server.
4 & 5: perform session closing.
6: `ws_sink.send(WsMessage::Close(None))` attempts to `write()` while the ws
   is no longer active. The error causes an early return, which means that
   the enqueued `Close` frame is not flushed.

Depending on the peer's shutdown sequence, this can result in the following
error, which can bubble up as a `Message` on the application's bus:

```
ERROR: from element /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0: GStreamer encountered a general stream error.
Additional debug info:
net/webrtc/src/webrtcsrc/imp.rs(625): gstrswebrtc::webrtcsrc:👿:BaseWebRTCSrc::connect_signaller::{{closure}}::{{closure}} (): /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0:
Signalling error: Error receiving: WebSocket protocol error: Connection reset without closing handshake
```

On the other hand, [`close()` ensures the ws is active] before attempting to
write a `Close` frame. If it's not, it only flushes the stream.

Thus, when we want to be able to close the websocket and/or to honor the closing
handshake in response to the peer `Close` message, the `ws_sink.close()`
variant is preferable.

This can be verified in the resulting server's logs:

```
tungstenite::protocol: Received close frame: None
tungstenite::protocol: Replying to close with Frame { header: FrameHeader { is_final: true, rsv1: false, rsv2: false, rsv3: false, opcode: Control(Close), mask: None}, payload: [] }
gst_plugin_webrtc_signalling::server: Received message Ok(Close(None))
gst_plugin_webrtc_signalling::server: connection closed: None this_id=192ed7ff-3b9d-45c5-be66-872cbe67d190
remove_peer{peer_id="192ed7ff-3b9d-45c5-be66-872cbe67d190"}: gst_plugin_webrtc_signalling::server: close time.busy=22.7µs time.idle=37.4µs
tungstenite::protocol: Sending pong/close
```

We now get the notification `Sending pong/close` (the closing handshake) instead
of `websocket start_send error` from step 6 with previous variant.

The `Connection reset without closing handshake` was not observed after this
change.

[send task]: 63b568f4a0/net/webrtc/signalling/src/server/mod.rs (L165)
[`WebSocket::close()` doc]: https://docs.rs/tungstenite/0.21.0/tungstenite/protocol/struct.WebSocket.html#method.close
[returns `SendAfterClosing`]: 85463b264e/src/protocol/mod.rs (L437)
[call to `do_close()`]: 85463b264e/src/protocol/mod.rs (L601)
[`close()` ensures the ws is active]: 85463b264e/src/protocol/mod.rs (L531)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1435>
2024-02-01 18:08:41 +01:00
Taruntej Kanakamalla
50e905fe4b webrtc: conditional compile for features with 1_22 dependency
Few features being used in webrtcsink like
the signal `request-aux-sender` are introduced
to webrtcbin in gstreamer release 1.22.

Rename the feature gst1_22 to v1_22 for uniformity.

Add v1_22 to default features.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1241>
2024-02-01 15:08:11 +05:30
Sebastian Dröge
f2a7a34abf rtp: gcc: Use x += ... instead of x = x + ... 2024-01-31 18:46:55 +02:00
Sebastian Dröge
a82068643d deny: Remove unnecessary toml_edit override
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sebastian Dröge
4ad101b53b Use once_cell crate directly again
The glib crate does not depend on it anymore and also does not re-export
it anymore.

Also switch some usages of OnceCell to OnceLock from std.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sebastian Dröge
08af298d11 gif: Update to gif 0.13
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sebastian Dröge
451d928026 webrtc: Update AWS signaller to http 1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sebastian Dröge
0e86dfa944 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 16:51:49 +02:00
Mathieu Duponchelle
ad51c61ac8 textwrap: add support for gaps
When accumulate-time is non-zero, we need to drain our accumulated
text once the threshold is reached.

Implement support for gaps the simplest way, by transforming it into
an empty buffer and chaining it through ourself.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1436>
2024-01-30 13:51:05 +01:00
Michael Tretter
4bb867bf52 livesync: add support for image formats
The livesync element is also useful for Motion JPEG streams. However,
Motion JPEG uses image/ caps instead of video/ caps.

The framerate is defined for image/, too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1440>
2024-01-29 11:07:30 +00:00
Michael Tretter
54f24fe4b0 meson: allow building plugins with GTK 4 examples
Only the examples of the fallbackswitch, livesync, and togglerecord
plugins require the gtk, gio, and gst-plugin-gtk4 features. The plugins
themselves don't actually have a dependency on GTK.

Only add the features (and examples) if the examples are actually
enabled to allow building these plugins without the GTK dependency.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1438>
2024-01-29 10:48:14 +00:00
Guillaume Desmottes
33a1d8de3d tracers: buffer-lateness: display some stats about late buffers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1437>
2024-01-29 09:24:08 +00:00
Guillaume Desmottes
d5740ea844 tracers: buffer-lateness: add argument to display only late buffers
Help to easily spot places where buffers are late when plotting big
pipelines.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1437>
2024-01-29 09:24:08 +00:00
Nirbheek Chauhan
5b0733d535 meson: Add nasm to PATH if meson can find it
Fixes rav1e build on Windows when built inside the monorepo.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1431>
2024-01-26 17:37:38 +00:00
Nirbheek Chauhan
6b79ce605d meson: pkg-config is required at build time
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1431>
2024-01-26 17:37:38 +00:00
Nirbheek Chauhan
8b5a398135 meson: Fix build on Windows with MSVC
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/480

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1431>
2024-01-26 17:37:38 +00:00
Michael Tretter
fcd57e9ac5 meson: remove trailing whitespace and add comma
Cleanup without functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1439>
2024-01-24 12:03:11 +01:00
Sanchayan Maity
95c007953c webrtchttp: Allow audio or video caps to be specified as None with WHEP
We were setting audio and video caps by default even when the user
might have requested only video or audio. This would then result
in a `Could not reuse transceiver` error from the webrtcbin.

Fix this by allowing the user to specify audio or video caps as
None. This allows us to maintain the earlier behaviour for backward
compatibility while allowing the user to not request audio or video
as need be.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1433>
2024-01-18 15:43:19 +05:30
Sebastian Dröge
764143d971 webrtc: Remove unnecessary manual Send+Sync implementations for signallers
These are automatically implemented.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/483

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-18 10:01:25 +02:00
Sebastian Dröge
1af18f3028 webrtc: Require Send+Sync for signaller implementations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-18 10:01:01 +02:00
Eva Pace
80b58f3b45 net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller
The JanusVRWebRTCSink is a new plugin that integrates with the Video
Room plugin of the Janus Gateway, which simplifies WebRTC communication.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
2024-01-17 20:33:57 +00:00
Maksym Khomenko
773ebc7854 webrtcsrc: don't restrict RTP extensions to TWCC only
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1381>
2024-01-17 07:34:01 +00:00
Guillaume Desmottes
c616423edb livesync: properly format jitter in debug logs
Easier to read that way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1430>
2024-01-16 13:46:34 +01:00
Sebastian Dröge
9556abb162 deny: Remove unnecessary overrides and add new one for livekit -> itertools
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
2024-01-16 07:52:48 +00:00
Sebastian Dröge
c8bd1293b9 inter: Update to serial_test 3
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
2024-01-16 07:52:48 +00:00
Sebastian Dröge
dfa95d8ed3 webrtc: Update to livekit-api / livekit-protocol 0.3
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
2024-01-16 07:52:48 +00:00
Sebastian Dröge
c85106e700 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
2024-01-16 07:52:48 +00:00
Maksym Khomenko
fecbe01e06 webrtcsink: make 'extensions' property usage conditional
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1423>
2024-01-16 07:13:56 +00:00
Sebastian Dröge
73a53e38c4 aws: s3: Disable remaining tests too for now
They fail state changes, which cases `GstHarness` to abort.
2024-01-16 09:13:41 +02:00
Arun Raghavan
fd3675aac0 aws: s3: Temporarily disable putobject tests
Disabling while we figure out why it's failing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1429>
2024-01-15 21:43:25 -05:00
Arun Raghavan
e70ef7f9e4 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
2024-01-11 15:38:57 -05:00
Arun Raghavan
8b18ca15b5 Revert "aws: Disable putobjectsink tests for now"
This reverts commit b128d127c2.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
2024-01-11 15:38:36 -05:00
Arun Raghavan
06213714c5 aws: putobjectsink: Fix a couple of minor log typos
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
2024-01-11 15:38:36 -05:00
Sebastian Dröge
fdd33fdeb0 deny: Remove a few de-duplicated dependencies
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1421>
2024-01-04 13:03:20 +02:00
Sebastian Dröge
cb78260d22 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1421>
2024-01-04 13:00:21 +02:00
Sebastian Dröge
d36c91d10f rav1e: Update to rav1e 0.7
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1421>
2024-01-04 12:59:50 +02:00
Nirbheek Chauhan
2d85048925 webrtc/signalling: We get the address when accepting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
2023-12-29 13:28:48 +00:00
Nirbheek Chauhan
63b568f4a0 webrtc/signalling: Fix potential hang and FD leak
If a peer connects via TCP and never initiates TLS, then the server
will get stuck in the accept loop. Spawn a task when accepting a TLS
connection, and timeout if it doesn't complete in 5 seconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
2023-12-29 13:28:48 +00:00
Guillaume Desmottes
d9397ef174 gtk4: fix build on Windows using winegl
from_glib_full() was not in scope.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1417>
2023-12-26 23:22:35 +01:00
Maksym Khomenko
17f0b61576 webrtcsink: add payloader-setup signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
2023-12-23 08:02:08 +00:00
Sebastian Dröge
1ef47cb48e Fix a few new clippy 1.75 warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1415>
2023-12-22 16:40:18 +02:00
Sebastian Dröge
79b8610fbe Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1414>
2023-12-22 15:46:51 +02:00
Sebastian Dröge
b128d127c2 aws: Disable putobjectsink tests for now
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
2023-12-22 13:25:12 +02:00
Sebastian Dröge
6686f6415f Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
2023-12-22 13:12:19 +02:00
Sebastian Dröge
df1f986239 Update plugin documentation cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
2023-12-22 11:41:01 +02:00
Arun Raghavan
06d96ec5a2 aws: Add plugin docs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 16:13:48 -05:00
Arun Raghavan
6d47045a60 aws: s3sink: Fix spelling of debug category
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 16:13:48 -05:00