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>
* A queue dedicated to transcript items not intended for translation.
* A queue dedicated to transcript items intended for translation. The items are
enqueued after a separator is detected or translate-lookahead was reached.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
This commit adds an optional experimental translation tokenization feature.
It can be activated using the `translation_src_%u` pads property
`tokenization-method`. For the moment, the feature is deactivated by default.
The Translate ws accepts '<span></span>' tags in the input and adds matching
tags in the output. When an 'id' is also provided as an attribute of the
'span', the matching output tag also uses this 'id'.
In the context of close captions, the 'id's are of little use. However, we can
take advantage of the spans in the output to identify translation chunks, which
more or less reflect the rythm of the input transcript.
This commit adds simples spans (no 'id') to the input Transcript Items and
parses the resulting spans in the translated output, assigning the timestamps
and durations sequentially from the input Transcript Items. Edge cases such as
absence of spans, nested spans were observed and are handled here. Similarly,
mismatches between the number of input and output items are taken care of by
some sort of reconcialiation.
Note that this is still experimental and requires further testings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
This commit adds an optional transcript translation feature implemented as
request src Pads.
When requesting a src Pad, the user can specify the translation language code
using Pad properties 'language-code'.
The following properties are defined on the Element:
- 'transcribe-latency': formerly 'latency', defines the expected latency for
the Transcribe webservice.
- 'translate-latency': defines the expected latency for the Translate
webservice.
- 'transcript-lookahead': maximum transcript duration to send to translation
when a transcript is hitting its deadline and no punctuation was found.
When the input and output languages are the same, only the 'transcribe-latency'
is used for the Pad. Otherwise, the resulting latency is the addition of
'transcribe-latency' and 'translate-latency'.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
We were currently returning a value based on the next chunk PTS, but the
expectation in GstAggregator is that we return a running time. This
resulted in spurious wakeups and warnings like:
0:00:01.501685123 1552995 0x55899715c1e0 WARN fmp4mux mux/fmp4/src/fmp4mux/imp.rs:1818:gstfmp4::fmp4mux:👿:FMP4Mux::drain_buffers:<fmp4mux0:sink_1> Don't have a complete GOP for the first stream on timeout in a live pipeline
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1127>
