This adds a check to change the capture settings to use 2 channels when
a channel number is encountered that would otherwise be interpreted as
SPEAKERS_UNKNOWN.
Because other capture methods may end up needing to share this code,
separate the window finding source code to window-helpers.c and
window-helpers.h.
This include a function to fill out a property list with windows, a
function to find a window based upon priority/title/class/exe, and a
function to decode the window title/class/exe strings from a window
setting string.
This adds code to set up the udev monitoring library and use the events
to detect connects/disconnects of devices.
When the currently used device is disconnected the plugin will stop
recording and clean up, so that the device node is freed up.
On reconnection of the device the plugin will use the event to
automatically start the capture again.
The graphics subsystem was not being freed here, for example if a
required effect failed to compile it would still successfully have the
graphics subsystem sans required effect. The graphics subsystem should
be completely shut down if required libobs effects fail to compile.
The remuxer thread was only started if there was an actual remux job,
which resulted in the remuxer thread not being able to call the worker's
destructor (because it wasn't running)
OBS Sparkle feeds have two extensions to vanilla Sparkle feeds:
- There can be two kinds of items per feed: (zipped) .app and .mpkg
via <ce:packageType>app|mpkg</ce:packageType> (default is mpkg)
- Feed items can be disabled via <ce:deployed>false</ce:deployed>; these
items will not be considered for updates unless
"[General] UpdateToUndeployed=1" is set the global config
Unlike other Sparkle implementations the FeedURL cannot be updated via user
preferences because we support multiple app packages with the same package
identifier but different FeedURL settings on the same machine
Due to a small error in the timestamp smoothing code the timestamp of
audio packages that were too early was always set to the next expected
timestamp, even if the difference was bigger than the smoothing threshold.
This would cause obs to simply append all audio data to the buffer even if
the real timestamp was way smaller than the next that was expected.
This should reduce corruption problems with for example the pulseaudio
plugin, which resends data under certain conditions.
If the sample format used by PulseAudio can not be converted into an
OBS audio format it will be handled as AUDIO_FORMAT_UNKNOWN which will
not result in a proper audio recording. So instead we request a format
that OBS supports from PulseAudio and let it do the format conversion.
The format PA_SAMPLE_S24_32LE is a 24 bit audio format in 32 bit integers
and not a 32 bit audio format and so it should no be mapped to
AUDIO_FORMAT_32BIT.