This change addresses a cosmetic issue with livekit, where the
connection quality indicator seen by other users shows bad quality
unless the track is added with a high quality layer. The details of the
layer submitted aren't significant for this purpose.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1457>
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state of the connection and [returns `SendAfterClosing`] if the socket is no
longer active, which is the case when a closing request has been received from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is no longer active.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/1457>
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/1457>
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/1410>
* Simplify state/playlist management
* Fix a bug that segment is not deleted if location contains directory
and playlist-root is unset
* Split playlist update routine into two steps, adding segment
to playlist and playlist write
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
This is required to take care of clock skew between
system time and pipeline time.
`track-pipeline-clock-for-pdt: true` mean utd time is
sampled for first segment and for subsequent segments
keep adding the time based on pipeline clock. difference
of segment duration and PDT time will match.
track-pipeline-clock-for-pdt: false` mean utd time is
sampled for each segment. system time may jump forward
or backward based on adjustments. If application needs
to synchronization of external events `false` is
recommended.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
- connect to `format-location-full` it provide the first
sample of the fragment. preserve the running-time of the
first sample in fragment.
- on fragment-close message, find the mapping of running-time
to UTC time.
- on each subsequent fragment, calculate the offset of the
running-time with first fragment and add offset to base
utc time
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
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/1388>
The "signaller" property used to be defined as MUTABLE_READY which meant that
the property was always set after `constructed()` was called.
Since `connect_signaller()` was called from `constructed()`, only the default
signaller was used.
This commit sets the "signaller" property as CONSTRUCT_ONLY. Using a builder,
this property will now be set before the call to `constructed()`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
During `on_remote_description_set()` processing, current session is removed
from the sessions `HashMap`. If an ice candidate is submitted to `handle_ice()`
by that time, the session can't be found and the candidate is ignored.
This commit wraps the Session in the sessions `HashMap` so an entry is kept
while `on_remote_description_set()` is running. Incoming candidates received by
`handle_ice()` will be processed immediately or enqueued and handled when the
session is restored by `on_remote_description_set()`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
