The reason to keep a reference counter for transitions is due to an
optimization I'm planning on when calculating transition volumes. I'm
planning on walking the source tree to be able to calculate the current
base volume of a source, but *only* if there are transitions active,
because the only time that the volume can be anything other than 1.0
or 0.0 is when there are active transitions, which may change the base
volume of a source.
When the presentation volume is set for a source, it's set for all of
its children and their children. The original intention for doing this
was to be able to use it for transitioning, but honestly it's just bad
design, and I feel there are better ways to handle transitioning volume.
Changed the design from using obs_source::enum_refs to just simply
preventing infinite source recursion in general, rather than allowing it
through the enum_refs variable. obs_source_add_child has been changed
so that it now returns a boolean, and if the function fails, it means
that the child cannot be added due to that potential recursion.
Two integers are needlessly converted to floating points for what should
be an integer operation. One of those floats is then used for another
integer operation later, where the original integer value should have
been used. So essentially there was an int -> float -> int conversion
going on, which could lead to potential loss of data due to floating
point precision.
There were also some general 64bit -> 32bit conversion warnings.
obs_encoder_getdisplayname declaration was not changed to match the
definition (obs_encoder_get_display_name) when the API consistency
update occurred.
On i3wm, windows aren't unmapped when switching away from a window's
workspace, but it does cause OBS to lose the capture. Because
switching back will not trigger a MapNotify, the capture fails to
restart unless you resize or move the window (ConfigureNotify). An
Expose event is fired by the wm, however, so catching this correctly
restarts the capture.
Refactor the screen enumeration code a little to make sure xinerama is present
and active before using it. If the extension is present but not active it will
no longer fail.
This adds support for the AverMedia C985 encoder (which is available on
C985 capture cards) as well as the C353 hardware encoder (which is
currently available on the X99S Gaming 9 motherboards).
These encoders have some limitations, such as limited resolutions
(1280x720 and 1024x768), a max GOP size of 30, and the encoder format
only supports YV12, which requires conversion if the current output
format isn't the same. The C985 and C353 encoders seem to be pretty
much identical, although it seems like the C353 has a bit more efficient
encoding.
I don't believe these are really suitable for streaming, as they do not
really have the encoding efficiency needed to stream at lower bitrates,
and seem to only support variable bitrate. However, for recording these
encoders are quite nice to have available, and work quite well.
The main module code was originally all packed in to the win-dshow.cpp
file, which isn't exactly ideal or clean if one wants to add other
things to the module as a whole.
Previously, due to a bug in libdshowcapture, the NV12 format was
actually being used for YV12 erroneously, and no actual support for YV12
existed. This fixes the bug with NV12 and adds support for YV12.
Waiting for the first packet to arrive before sending the headers helps
prevent issues with certain types of encoders that may not get their
header/SEI until the first packet has been received.
If an encoder did not possess any SEI data, it would never send data at
all because the sent_first_packet wasn't set despite the first packet
being sent.
Added obs_avc_keyframe that returns whether an avc packet is a keyframe
or not. This function is particularly useful for when writing custom
encoder plugins.
I encountered some cases where I needed to use these enumerations
outside of the file, so this allows other modules to use AVC
enumerations without having to redefine them each time. Especially
useful for custom encoder modules.
I neglected to surround some files with extern "C", so if something
written with C++ used the files it would cause function exports to not
be mangled by it correctly.
This adds bicubic and lanczos scaling capability to libobs to improve
scaling quality and sharpness when the output resolution has to be
scaled relative to the base resolution. Bilinear is also available,
although bilinear has rather poor quality and causes scaling to appear
blurry.
If the output resolution is close to the base resolution, then bilinear
is used instead as an optimization, as there's no need to use these
shaders if scaling is not in use.
The Bicubic and Lanczos effects are also exposed via exported function
to allow the ability to use those shaders in plugin modules if desired.
The API change adds a variable 'scale_type' to the obs_video_info
structure that allows the user interface to choose what type of scaling
filter should be used.
Copying this binary is kind of unnecessary because it's something that
everyone has as long as they update DirectX, and isn't something we can
distribute because it's a Microsoft DLL.
Remove the update time check from the volume meter since it is no longer
nescessary. The update interval can now be set with the volmeter which has
the added benefit of not loosing data where previously updates had been
skipped.
Remove the calculation of volume levels and the corresponding signal
from obs_source since this is now handled in the volume meter.
Code that is interested in the volume levels can either use the
volmeter provided from obs_audio_controls or use the audio_data signal
to gain access to the raw audio data.
Signal updated volume levels when they become available in the volume
meter. The frequency of the updates can be adjusted by setting a
different update interval.
Remove the the signal handler for the volume_level signal of audio
sources from the volume meter in anticipation of using the levels
calculated in the volume meter itself.
Add a property to the volume meter that specifies the length of the
interval in which the audio data should be sampled before the
audio_levels signal is emitted.
jp9000
skwerlman
fryshorts
Emil Sayahi
raincomplex
Jake Probst
Zachary Lund