Contrary to what one might believe, this actually reduces the size of
the structs due to alignment constraints. On Linux x86-64 clang/gcc it
reduces the size of the caption_frame_t struct from 7760 bytes to 6800
bytes, on Windows x86-64 MSVC from 11600 bytes to 6800 bytes.
It also causes simpler and potentially faster assembly to be generated
as the values can be directly accessed as uint8_t instead of having to
extract the corresponding bits with bitwise operations.
It also gives us the same ABI with clang/gcc and MSVC.
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.
