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>
This is no longer available as this could lead to building a defined
value in Rust which could be interpreted as undefined in C due to
the sentinel `u64::MAX` for `None`.
Use the constants (e.g. `ONE`, `K`, `M`, ...) and operations to build
a value and deref (`*`) to get the quantity as an integer.
In roll-up mode, when no more timed text comes in, the closed
captions may remain displayed on screen indefinitely (unless the
decoder implements a timeout, but that is not mandatory).
Expose a property to erase the display memory after a configurable
amount of time has elapsed instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/754>
There can be small race where transcription-bin is linked with
tee but state change of the transcription-bin is not finished.
And at the same time, upstream pushes event/buffer to the
transcription-bin. Do state change first then link to avoid
the condition
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/716>
Zero-padding is not specified for the indices but all time components
need to be zero-padded (3 digits for fractional seconds, 2 digits for
everything else).
If transcription runs slow or has issues the queue can fill up and block
all audio processing. This gives the queue a sufficent buffer and allows
it to drop audio if it eventually fills up. This was most noticable with
bad internet connections using the `awstrnascriber` where it would take
quite a while for the websocket to eventually timeout and the bin to
enter `passthrough=true`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/688>
By using this new property, application can select exclusive caption
source. There are three source types
- Both: Inband and transcription captions are combined if exist.
This is default behavior.
- Inband: Transcription buffers will be dropped
- Transcription: Caption meta of each video buffer will be dropped
In this version, transcriberbin doesn't provide any hint
for application to help caption source decision. That can be done
by application's strategy, passthrough status or probing inband
caption meta for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/684>
Fix race between latency query handler and setup_transcription()
method.
Locking order of setup_transcription() is
state lock -> setup_transcription() -> settings lock
So taking state lock inside of setting lock in src_query()
can cause deadlock.
I hadn't really tested the element with pop-on mode, and the row
for each line in the input text was hardcoded to 13, which was
clearly wrong.
Switch to incrementing it properly.
C.9 Automatic Caption Erasure (Preferred)
[...]
Some manufacturers have suggested building automatic timeout into their
decoders. They propose that if no data are received for the selected caption
channel within a given time, the decoder should automatically erase the
caption. Such erasure may supersede the intentions of the caption service
providers and institute one maximum display time for all captioning services.
If such a timeout is deemed necessary, however, the time limit should be no less
than 16 seconds, an amount of time said by caption service providers to be longer
than their most enduring caption. It is preferred, when automatic caption erasure
is used in a decoder, that only displayed memory be erased, since some caption
service providers may, contrary to recommended practice (see Section B.8.3), send
pop-on style caption data to non-displayed memory more than 16 seconds before
sending the EOC command which causes the caption to display.
In this mode, cues are output as soon as they are ready for
display, without a duration. This can be useful in live mode,
when downstream is OK with determining the duration after the
fact, through clear=True.
The consequence of this is that the current roll-up window will
be output repetitively, it is up to downstream to deal with that
how it prefers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/554>
There is no point to that, the code is already factored in such
a way that erase_display_memory is inserted at the correct time,
including while loading the next pop-on captions in non displayed
memory.
Locking order of state and settings was inconsistent, and causing
deadlocks. Fix and document it, consistently drop locks before
chaining up events / pushing and avoid sequentially unlocking /
relocking settings in the same local code path.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/539>
This new element puts together some of the elements we've written
in recent times (awstranscriber, tttocea608, textwrap, cccombiner)
into a convenience high-level element.
The design of the element is AV in -> AV (+ CC metas) out.
The element exposes property to set and unset a "passthrough" mode,
during which the transcriber element's state is set to NULL but kept
in the bin, in order for the user to be able to set properties on
sub elements no matter what the current mode is, using the
GstChildProxy interface.
In addition, the element ensures that the latency it reports stays
fixed so that playback continues uninterrupted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/528>
As stated in the spec:
> In addition, the user must have the capability to select a black
> background over which the captioned letters are displaced.
The property is MUTABLE_PLAYING
In roll-up modes, we open new lines when the last column is reached.
This commit implements lookahead on a word basis, in order to avoid
splitting words unless absolutely necessary (when a word won't fit
on a full row)
