Details in #6226, basically this fixes an issue where OBS links
to the system-installed librtmp1, which is incompatible with our
librtmp.
Co-authored-by: ewhac <ewhac@ewhac.org>
Due to the way the frontend works, the logic for inserting the row for
the group in the UI was dropped since the new item_add message will
insert one automatically since it forces a refresh of the scene items.
The reliance on that implicit row insertion when grouping items is a
bit of a code smell, although the alternative would be to add logic to
disable handling the signal for that instant, which is probably a
worse choice.
Allows a frontend the ability to set the maximum audio buffering
latency, and specify whether that audio buffering is either fixed (to
the maximum audio buffering latency), or dynamically increasing from 0.
This will be useful if the user wishes to output audio to devices or
through a virtual audio device at a guaranteed minimal latency.
When audio buffering is maxed, certain sources will always repeat the
same debug logging message of "render audio source [name] has gone
backwards", which shouldn't apply if the audio timestamp is 0 rather
than a valid audio value.
This is in anticipation to adding low audio latency mode, will cause
audio buffering to be fixed rather than dynamically increasing. Having
fixed audio buffering means that audio latency always stays the same
rather than dynamically increasing.
The Windows build script would respect the user-specified build
directory for the configure step, but not for the build step. Fix that
by applying the same logic in Configure-OBS to Build-OBS.
This removes the min and max size properties from the YouTube "Manage
Broadcast" dialog. Such values depend on the actual content or very
precise layout management, making hardcoded values prone to being
incorrect. To avoid incorrect values we should avoid these settings on
top level widgets/windows.
This pattern can lead to crashes on Wayland compositors like
GNOME which are strict about these values being correct.
Timestamp of some of audio sources desync over the time and audio glitch
is suspected to be caused when the difference between `timestamp` and
`next_audio_ts_min` exceeds TS_SMOOTHING_THRESHOLD.
When such condition happens, leave a log message to investigate the
glitch of the audio.
Want to avoid receiver gamut remapping from Rec. 2100 to P3.
Fix min luminance appearing as NaN by switching from 0/0 to 0/1.
Also set max luminance for HLG because why not.
Defaulting to 1 means that it can't be differentiated between whether a
source really has 1 channel. Now, should a source not have any audio
(yet), this function makes it very clear that this is the case by
returning 0.
