There is some broken software out there not inserting the empty lines
and we don't really need them for proper parsing. Only require an empty
line between header and the first caption line.
This expands the depth input check into proper validation and resulting
enum, which can be reused by the rest of the logic.
Signed-off-by: Luca BRUNO <lucab@lucabruno.net>
We don't want to run it every time a strong reference is dropped but
only at the very end. Otherwise dropping the socket stream will cause a
panic because the socket itself is still running.
This can only happen if the receiver is dropped, which only happens when
the task is stopped. As such, Flushing should be returned instead of
panicking.
This should start making navigating the tree a little easier to start
with, and we can then move to allowing building specific groups of
plugins as well.
The plugins are moved into the following hierarchy:
audio
/ gst-plugin-audiofx
/ gst-plugin-claxon
/ gst-plugin-csound
/ gst-plugin-lewton
generic
/ gst-plugin-file
/ gst-plugin-sodium
/ gst-plugin-threadshare
net
/ gst-plugin-reqwest
/ gst-plugin-rusoto
utils
/ gst-plugin-fallbackswitch
/ gst-plugin-togglerecord
video
/ gst-plugin-cdg
/ gst-plugin-closedcaption
/ gst-plugin-dav1d
/ gst-plugin-flv
/ gst-plugin-gif
/ gst-plugin-rav1e
gst-plugin-tutorial
gst-plugin-version-helper
This normalizes the loudness of an audio stream to a target loudness
with a given maximum peak based on EBU R128.
Conceptually it keeps a 3s lookahead for calculating the perceived
loudness and based on that calculates the gain required to reach the
target loudness. The calculated gains then go through a gaussian filter
for smoothening and are then applied to the audio in 100ms blocks. Each
of the 100ms blocks is then passed to a limiter filter to prevent going
above the maximum peak.
See http://k.ylo.ph/2016/04/04/loudnorm.html for some more details about
the algorithm.
It introduces 3s of latency and currently only works on 192kHz audio.
Using it with a different sample rate requires resampling before and
afterwards. The upsampling is required to calculate the true peak.
Other than the ffmpeg filter it currently does not support two-pass
processing but only one-pass/live processing.
Compared to the ffmpeg filter this code was refactored considerably and
the limiter implementation was fixed to actually work, as well as
various other bugs in different places that were fixed.
This commit fixes several issues with the `Ts*Src` elements.
The pause functions used cancel_task which breaks the Task loop at await
points. For some elements, this implies making sure no item is being lost.
Moreover, cancelling the Task also cancels downstream processing, which
makes it difficult to ensure elements can handle all cases.
This commit reimplements Task::pause which allows completing the running
loop iteration before pausing the loop.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/277#note_439529
In the Paused state, incoming items were rejected by TsAppSrc and DataQueue.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/277#note_438455
- FlushStart must engage items rejection and cancel the Task.
- FlushStop must purge the internal stream & accept items again.
If the task was cancelled, `push_prelude` could set `need_initial_events`
to `true` when the events weren't actually pushed yet.
TsAppSrc used to renew its internal channel which could cause Buffer loss
when transitionning Playing -> Paused -> Playing.
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/issues/98
