Enable B-Pyramid frames to improve quality for high motion content.
B-pyramid allows a B-frame to choose closer frames for reference which
may have higher correlation.
Adjusts packet priority to avoid dropping referable B-frames
A check has been added to ensure feature is enabled only on platforms
with MSDK API 1.8 or higher - addresses crashing issues.
Unlike get_properties, there is not reason to not call get_defaults if it is
given in addition to get_defaults2. Additonally this fixes the bug with
'init_encoder' which would only ever call get_defaults, resulting in broken
encoders if those used get_defaults2.
Move xlock after initial window listing
- XErrorLock uses static fields so when you use them recursively
they collide and do weird things. This keeps the two xlocks from
colliding.
Build obs textures based on returned pixmap texture types to ensure
compatibility during copies. (prevents glCopyImageSubData failed due to
texture incompatibilities)
Fix alpha check by using pixmap depth instead of checking channel sizes
(wrong on nvidia and maybe amd hardware)
Fix X error checking, now you will get 1 BadAlloc when pixmap allocation
fails instead "glCopyImageSubData failed" every frame. (occurs on Gnome
under special circumstances)
Adds support for pausing recordings. When settings are eligible for
recordings, a pause button will appear next to the recording button. If
the settings are not eligible, it will warn the user in the output
settings that they cannot pause recordings if those settings are used.
This implements pausing of outputs. To accomplish this, raw audio/video
data is halted to the encoders or raw output. Pausing is as precisely
timed as possible according to the timing of the obs_output_pause call,
and audio data will be spliced down to the exact audio sample in
accordance to that timing at the start/end marks.
Outputs that support this (outputs used for recording) can set the
OBS_OUTPUT_CAN_PAUSE capability flag.
If the audio subsystem was buffered to any extent, the audio of a raw
output would start off at a negative offset, requiring each raw output
to implement a "prepare_audio" function (as seen in the FFmpeg output)
in order to ensure proper synchronization with video. This did not
apply to encoded outputs because it was already being performed by the
obs-encoder code.
If an audio source does not provide enough data at a steady pace, the
timestamp update does not happen, and buffering increases until it
maxes out. To counteract this, update the timestamp anyway.
Another issue for decoupled audio sources is that timing is not
adjusted for divergence from system time. Making this adjustment is
better for timing stability.
5+ hours of stable audio without any buffering on my GV-USB2 where it
used to add 21ms every 5 mintues or so.
Fixes https://obsproject.com/mantis/view.php?id=1269
If a device produces video and audio timestamps atdifferent rates,
this divergence can cause massive buffering on the audio side, leading
to a capped audio buffer, and total sound loss. This change allows a
hardcoded list of devices to use the existing decoupling logic. For
now, only "GV-USB2" has been added.
When combined with another fix, 5+ hours of stable audio without any
buffering on my GV-USB2 where it used to drop sound completely after
an hour or so.
Partially fixes https://obsproject.com/mantis/view.php?id=1269
Fix ternary test to use BGRX render targets for YUV to RGB
conversions. The previous behavior may have been fine though since
the shaders fill the alpha channel with 1.0 anyway.