* 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>
This helps gather together the details related to the `TranscriberLoop`.
One difference with previous implementation is that the ws `Client` is
build each time the loop is started instead of being reused. With the new
approach, we don't keep the connection open after EOS and we should be
more resistant in case of a connection failure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
Instead of sending transcription events to the src pad loop, this commit
enqueues the transcribed buffers immediately in the ws loop, then notifies
the src pad loop. The src pad loop is only in charge of dequeuing the buffers.
This should help with upcoming evolutions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
If creating a playlist or fragment stream fails (disk is full, the
directory is removed, ...), we will currently crash because the signal
handler expects a non-None GIOStream. The actual callback is allowed to
return None values and we handle this in the caller, so let's not have
this restriction on the signal handler.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1093>
for uploaded object default content-type is set to binary/octet-stream,
which is correct.
metadata cannot be used to set content-type and content-disposition as
setting metadata add a prefix x-amz-meta to key
e.g. setting metadate "content-type=video/mp4" actually set value as
x-amz-meta-content-type. So these has to be seaprate property.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1085>
Commit ad3f1cf fixed the name of hlssink child element to be the same
for hlssink2 and hlssink3. However, we rely on element name to return
boolean in case of hlssink3 or None in case of hlssink2 as the return
value of the delete-fragment closure.
Fix this by using the factory name instead of the element name.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1076>
Simplifies state tracking and potentially reduces latency as it's not
necessary to wait until all fragments of an OBU are received.
The last OBU of a TU is marked with the marker flag to allow parsers to
detect this without first seeing the beginning of the next TU.
Also use a simple `Vec` for collecting complete OBUs instead of a
`gst_base::Adapter` as this reduces the number of allocations.
And also handle invalid packets a little bit more gracefully.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/244
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
We dispose of consumer pipelines asynchronously, potentially after the
session objects have been disposed of.
As session objects are the owner of the cc element, it is entirely
possible for the bwe-request signal to get emitted after cc has been
disposed of, as the closure only takes a weak reference to it.
Fix by simply checking if cc is None
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1044>
Without this auto-pluggers such as decodebin or parsebin will be unable to
process AV1 RTP payloads.
Tested with: `videotestsrc num-buffers=50 ! videoconvert ! av1enc ! av1parse ! rtpav1pay ! queue ! decodebin3 ! videoconvert ! queue ! autovideosink`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1034>
Right now the code manually pieces together the components
in a String for efficiency. When credentials contain special
characters this can result in invalid URLs, so do it the proper
way (with Url::parse + format) to make sure components are escaped
as needed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
WHIP endpoint providers like Cloudflare do not support Trickle ICE
and need candidates to be send along with the initial offer. Instead
of sending the offer in create-offer promise, send it once the ICE
candidates have been gathered.
While at it add properties to set STUN and TURN server along with the
ICE transport policy as at least when testing the Cloudflare WHIP
endpoint seems unreachable without it. This has also been observed
with Cloudflare provided demos.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
This implements WHEP specification based on
https://datatracker.ietf.org/doc/html/draft-murillo-whep-00
and has been tested with Cloudflare.
Server offers are likely to be removed from the WHEP specification
in upcoming revisions, to avoid compatibility issues. None of the
commercial services implementing WHEP support server initiated offers.
So we only support client side initiated offers.
Follows session setup and tear down as covered in Figure 1, Section 3
of the specification.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
