The internal (C) jitterbuffer needs to know about the configured
latency when calculating a PTS, as it otherwise may consider that
the packet is too late, trigger a resync and cause the element to
discard the packet altogether.
I could not identify when this was broken, but the net effect was
that in the current state, ts-jitterbuffer was discarding up to
half of all the incoming packets.
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>
Writing a proper "depfile" to follow depending files, based on depfiles
generated by rustc.
This is based on work done while working on gobject-examples-rs
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>
I give up on crossbeam_channel. For some reasons some receivers are not
always unblocked and I was not able to reproduce using simpler test
cases.
Use with mpsc channels instead which are more reliable.
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).
Previous version used the Context::block_on_or_add_sub_task which
spawns a full-fledged executor with timer and io Reactor for no
reason when we just need to wait for a Receiver or JoinHandle.
When the iteration loop is throttling, the call to `abort` on the
`loop_abort_handle` returns immediately, but the actual `Future`
for the iteration loop is aborted only when the scheduler throttling
completes. State transitions which requires the loop to be aborted &
which are serialized at the pipeline level can incur long delays.
This commit makes sure the Task Context's scheduler is awaken as soon
as the task loop is aborted.
Keep the state mutex during the whole decodebin pad-added callback.
Fix a race when we were checking if state.waiting_for_ss_eos was set and
it was removed before we actually processed the item.
Fix#184
This implements a default timeout and retry duration for the remaining
S3 requests that were still able to be blocked indefinitely. There are 3
classes of operations: multipart upload creation/abort (should not take
too long), uploads (duration depends on part size), multipart upload
completion (can take several minutes according to documentation).
We currently only expose the part upload times as configurable, and hard
code the rest. If it seems sensible, we can expose the other two sets of
parameters as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/690>
Previously, the actual reading from the streaming body of a GetObject
request was not within the same timeout/retry path as the dispatch of
the HTTP request itself. We consolidate these two into a single async
block and create a sum type to encapsulate the rusoto and std library
error paths within that future.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/690>
It might still be blocked downstream for a while, e.g. in the clocksync.
Flushing does not cause any problems as fallbackswitch is not going to
forward it and will only unblock everything up to there.
Have seen a few times where machines that are in perfect time sync with a good source the requests fail with `RequestExpired` errors.
https://docs.aws.amazon.com/transcribe/latest/dg/CommonErrors.html
While not perfect, bumping to five minutes gives more a chance that the signed requests to start streaming won't be expired.
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>
