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>
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>
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>
If a basicna character is received, it will always have a channel of 0
even if it's directed at a different data channel. Fix by keeping track
of the last channel from other commands and using that when producing
text in the basicna subset.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1112>
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>
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>
With this, if the transcriber element in use supports "translation_src_"
request source pads, the user can now specify what languages to
translate to and how to map them to 608 channels (only CC1 and CC3 are
supported).
For instance, translation-languages="languages, CC3=transcript, CC1=fr"
will cause the original transcript to be muxed into the CC3 channel, and
the French translation to be muxed into the CC1 channel.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1149>
To avoid special characters getting de-duplicated by the decoder, we
insert no-op control commands after those. The no-op command must be
picked according to the mode we're in however, inserting
"resume_caption_loading" commands in roll-up mode caused obvious issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1147>
Allowed downstream caps might hold multiple structures, simply fixating
the first structure is not enough, tttocea608 must also create caps with
a single structure from there (or remove the remaining structures, but
new caps seems cleaner)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1146>
When passthrough=false at construction and the transcription bin
is linked after receiving video caps (and not on state change),
there could be a race where transcription-bin was linked with
tee but state change of the transcription-bin was not finished.
If upstream pushed a buffer at that point, it got a flushing flow
return and stopped streaming.
This is the same issue and the same fix as 558656deb5
for the initial passthrough=false case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1142>
