Commit graph

2674 commits

Author SHA1 Message Date
Sebastian Dröge
d0ea7587d2 Update GStreamer bindings in Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1456>
2024-02-12 19:07:07 +02:00
Sebastian Dröge
289fdaf490 gtk4: Always draw a black background behind the video frame
This makes sure that there is the same background behind the frame, no
matter if black borders have to be added or not. Without this a frame
that has transparency would change rendering depending on the layout.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1456>
2024-02-12 19:07:07 +02:00
Ruben Gonzalez
ea93b8896d gtk4: Fix segfault running gst-inspect -a when GTK4 and GTK3 is installed
Segmentation fault when getting default value of paintable property
from gtk4paintablesink element when libgtk-4.so.1 from libgstgtk4.so
and libgtk-3.so.0 from libgstgtk.so are installed:

> cannot register existing type 'GdkDisplayManager'

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/490

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1456>
2024-02-12 19:07:07 +02:00
Mathieu Duponchelle
32cb2877fd 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/1456>
2024-02-12 19:07:07 +02:00
Michael Tretter
2d796e5108 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/1456>
2024-02-12 19:01:42 +02:00
Michael Tretter
0f1fde5d0a 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/1456>
2024-02-12 19:01:42 +02:00
Nirbheek Chauhan
43b9aab88e 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/1456>
2024-02-12 19:01:42 +02:00
Nirbheek Chauhan
de64699eaa meson: pkg-config is required at build time
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1456>
2024-02-12 19:01:42 +02:00
Nirbheek Chauhan
e782fd14c2 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/1456>
2024-02-12 19:01:42 +02:00
Michael Tretter
3bc6f7c614 meson: remove trailing whitespace and add comma
Cleanup without functional change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1456>
2024-02-12 19:01:42 +02:00
Guillaume Desmottes
90342ff90e livesync: properly format jitter in debug logs
Easier to read that way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1456>
2024-02-12 19:01:42 +02:00
François Laignel
0e9d33b38b 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/1456>
2024-02-12 17:50:52 +01:00
François Laignel
231389a990 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/1456>
2024-02-12 17:50:24 +01:00
Nirbheek Chauhan
3cfb28d048 webrtc/signalling: We get the address when accepting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1456>
2024-02-12 16:56:02 +02:00
Nirbheek Chauhan
e3190c888a 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/1456>
2024-02-12 16:53:06 +02:00
Sebastian Dröge
c03e108487 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1411>
2023-12-18 10:42:48 +02:00
Arun Raghavan
7bb277884b threadshare: Fix a deadlock in used-socket notification
This manifests in a gst-launch-1.0 pipeline using ts-udpsrc, since
notification of used-socket results in the property being read by the
application, and the settings lock causes a deadlock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1411>
2023-12-18 10:27:16 +02:00
Arun Raghavan
9d802b1969 threadshare: Fix a typo while logging
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1411>
2023-12-18 10:27:16 +02:00
Mathieu Duponchelle
630a1120ba webrtcsink: don't panic on failure to request pad from webrtcbin
webrtcbin will refuse pad requests for all sorts of reasons, and should
be logging an error when doing so, simply post an error message and let
the application deal with it, the reason for the refusal should
hopefully be available in the logs to the user.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1411>
2023-12-18 10:27:16 +02:00
Seungha Yang
55d2b9483c fallbacksrc: Fix timeout scheduling
Other thread can schedule the timeout (e.g., unblock signal
or active pad change) while state lock is released

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1411>
2023-12-18 10:25:50 +02:00
Sebastian Dröge
2136befbef ndi: Don't mark private type as public
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1411>
2023-12-18 10:24:32 +02:00
Sebastian Dröge
11ea962f81 ndi: Remove wrong Clone impl on RecvInstance
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1411>
2023-12-18 10:24:32 +02:00
Tim-Philipp Müller
b81b2cee81 net: aws: tests: use unicode escapes for unicode characters
.. instead of putting actual multi-byte unicode characters into
the source code which appears to make hotdoc unhappy.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1390>
2023-11-13 12:58:25 +00:00
Sebastian Dröge
ee3a272859 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Sebastian Dröge
f5d633d293 Update versions to 0.9.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Sebastian Dröge
4e05fc5999 sccparse: Fix leading spaces between the tab and caption data
CCExtractor is creating files like this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Sebastian Dröge
eff3618ebd Set sync=false in rsfilesink / s3sink
BaseSink defaults to sync=true and that doesn't make much sense for
these elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
9ffcb14617 livesync: Remove the stop from outgoing segments
Our buffer duplication can extend a segment indefinitely.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/452
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
8114f94f73 livesync: Keep existing buffer duration in some cases
Resize a repeat buffer only if caps gave us a duration to use, or we
consider its current duration unreasonable.

In particular, for audio streams we should prefer reusing the buffer
size upstream gave us, as we did before 6633cc4046.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
72506b94e3 livesync: Split fallback_duration into in_ and out_duration
Make it independent of the `latency`; this was inconsistent anyway,
where the default latency of zero got you a fallback duration of 100 ms
and something else got you half the latency.

Maintain a separate duration for the `in` and the `out` side so we
change the duration of repeat buffers after a caps change, not just
before.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Guillaume Desmottes
dafdb48ff6 livesync: display jitter when waiting on clock
We already log the result of the clock wait call so may as well log the
returned jitter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Guillaume Desmottes
4f9d5b7a79 livesync: log new pending segments
The debug print of the event does not display details about the segment:
  Unqueueing Some(Event(Event { ptr: 0x7fa3e0002580, type: "segment", seqnum: Seqnum(479), structure: Some(GstEventSegment { segment: (GstSegment) ((GstSegment*) 0x7fa3e8001d00) }) }))

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
1eaac4e0a2 livesync: example: Add identities single-segment=1
These let us change the runtime offset of the test buffers via pad
offsets without pushing new segments into livesync, which is necessary
to demo the late-threshold behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
b8e891b5b7 livesync: Use fallback_duration for audio repeat buffers as well
Don't depend on upstream giving us sanely-sized buffers if we want to
repeat.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
474f00f8d6 livesync: Separate out_buffer duplicate status from GAP flag
Otherwise we might get confused by upstream GAP buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
98d839a920 livesync: Handle flags and late buffer patching after queueing
This makes the chain function almost independent of the output state. We
still do the early discard check with `buffer_is_backwards` so we don't
try to queue buffers we can't use, allowing us to fast-forward upstream
without blocking on the src task.

Don't accept `LateOverThreshold` buffers when we have `pending_caps` or
a `pending_segment`. We need to apply these first before we can sensibly
patch buffers from the new stream.

Deduplicate most of the output buffer patching code into a new
`patch_output_buffer` method.

For: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/450
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
f565875b6c livesync: Simplify num_duplicate counting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
873b91d7d5 livesync: Move num_in counting to the src task
This is in preparation for moving more accept/discard logic to the src
task, so we can only count `num_in` here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
395ef40fc9 livesync: Move a notify closer to the interesting state change
Move the `notify_all` to where we pop the buffer. We're moving within a
single state lock so no change in behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
ad97b21011 livesync: Replace an if-let with match
No change in behavior, yet. Separate commit to ease reviewing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
fa05a305b6 livesync: Clean up state handling
- Separate resetting state more cleanly, introducing `set_flushing`,
  `sink_reset` and `src_reset`.
- Clear the queue early when we flush, in order to unblock waits on
  query responses.
- Return an error when we fail to start, pause or stop the task.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
274e3cd71b livesync: Log a category error when we are missing the segment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
ab84da6d32 livesync: Improve audio duration fixups
- An entirely missing duration is now only logged at debug level instead
  of pretending the duration was zero and warning about it.
- Silently fix up a duration difference up to one sample.
- Error when we fail to calculate the duration; don't try to apply the
  `fallback_duration` to a non-video stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
7a70863152 livesync: Simplify start_src_task and src_loop
This should effect no change in behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Jan Alexander Steffens (heftig)
4c4aff3a5b livesync: Rename activatemode methods to *_activatemode
This matches the other plugins.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Arun Raghavan
d35fc2eb6d aws: s3: Properly percent-decode GstS3Url
We previously only percent-decoded the first fragment. This doesn't
necessarily harm anything, but for consistency we keep the structure
un-encoded, and encode when converting to a string representation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Arun Raghavan
8f97d691e1 aws: s3sink: Fix handling of special characters in key
Properly URL-encode the string if needed, and add some tests for a
couple of cases.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/431
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Sebastian Dröge
7d2e849bbe rtpav1depay: Don't push stale temporal delimiters downstream
Only push them downstream once a complete OBU was assembled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Sebastian Dröge
7bb6f0265c rtpav1depay: Skip unexpected leading fragments
If a packet is starting with a leading fragment but we do not expect to
receive one, then skip over it to the next OBU.

Not doing so would cause parsing of the middle of an OBU, which would
most likely fail and cause unnecessary warning messages about a
corrupted stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00
Lieven Paulissen
beb524e950 ndisrc: Assume input with more than 8 raw audio channels is unpositioned
gst_audio_channel_positions_from_mask() will otherwise print warnings
all the time.

Fixes #444

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:47:41 +02:00