This allows us to change the visible UI name of a property after it's
been created (particularly for a case where I want to change an
'Activate' button to 'Deactivate')
There appears to be a bug with displaying the vertical scroll bar widget
where the horizontal scroll bar will show when it's not supposed to.
Fortunately it can be completely disabled.
The regular scroll area can expand horizontally, but the problem with
this is that sometimes there are controls within it that expand way too
big.
For example, the properties window for window capture can have a list of
windows where the titles of the windows are really really long, and it
causes the properties to extend way too far to the right, making the
window look really unusual.
Another example are the volume controls in the main window that can
expand way to the right if the name of a source is really long, causing
the volume control to stretch way too far to the right, making the
volume controls difficult to use when that happens.
So this just makes it so it sets the maximum width of a scroll area's
internal widget to the actual width of the scroll area, preventing it
from going off the side of the scroll area.
This crash report dialog is mostly just for the windows crash handling
code. If a crash occurs, the user will be able to view the crash report
and post it on the forums or give it to a developer for debugging
purposes.
A slightly refactored version of R1CH's crash handler, allows crash
handling for windows which provides stack traces of all threads and a
list of all loaded modules. Also shows the processor, windows version,
and current libobs version.
I originally had it set the color space and color range in the video
info callback, but I forgot that it's a function that's called after the
encoder is initialized. You can change the color space and color range,
but you have to reconfigure the encoder, and there's no real reason to
do that.
When the encoder is set to scale to a different resolution than the obs
output resolution, make sure it uses the current video colorspace and
range by default.
Uses the output duplicator API in order to get a high performance
monitor capture on windows 8+. This is actually designed to be
interchangeable with regular GDI-based monitor capture (uses the same
source id).
Allow the user to select whether to buffer the source or not. The
settings are auto-detect, on, and off. Auto-Detect turns it off for
non-encoded devices, and on for encoded devices.
Webcams, internal devices, and other such things on windows do not
really need to be buffered, and buffering incurs a tiny bit of delay, so
turning off buffering is actually a little better for non-encoded
devices.
I actually kind of hate how strstr returns a non-const even though it
takes a const parameter, but I can understand why they made it that way.
They really should have split it in to two functions though, one const
and one non-const or something. But alas, ultimately for a C programmer
who knows what they're doing it isn't a huge deal.
This adds support for the windows 8+ output duplicator feature which
allows the efficient capturing of a specific monitor connected to the
currently used device.
Suppress signals of the volume input when setting a value. This stops
the volume control from setting the source volume when it receives the
volume changed event from the source.
Suppress signals of the volume slider when setting a value. This stops
the volume control from setting the source volume when it receives the
volume changed event from the source.
Use a dedicated method for setting the dB label in the volume control
and make sure to call it regardless if the volume was changed through
the slider or from somewhere else.
Previously, the design for the interaction between the encoder thread
and the graphics thread was that the encoder thread would signal to the
graphics thread when to start drawing each frame. The original idea
behind this was to prevent mutually cascading stalls of encoding or
graphics rendering (i.e., if rendering took too long, then encoding
would have to catch up, then rendering would have to catch up again, and
so on, cascading upon each other). The ultimate goal was to prevent
encoding from impacting graphics and vise versa.
However, eventually it was realized that there were some fundamental
flaws with this design.
1. Stray frame duplication. You could not guarantee that a frame would
render on time, so sometimes frames would unintentionally be lost if
there was any sort of minor hiccup or if the thread took too long to
be scheduled I'm guessing.
2. Frame timing in the rendering thread was less accurate. The only
place where frame timing was accurate was in the encoder thread, and
the graphics thread was at the whim of thread scheduling. On higher
end computers it was typically fine, but it was just generally not
guaranteed that a frame would be rendered when it was supposed to be
rendered.
So the solution (originally proposed by r1ch and paibox) is to instead
keep the encoding and graphics threads separate as usual, but instead of
the encoder thread controlling the graphics thread, the graphics thread
now controls the encoder thread. The encoder thread keeps a limited
cache of frames, then the graphics thread copies frames in to the cache
and increments a semaphore to schedule the encoder thread to encode that
data.
In the cache, each frame has an encode counter. If the frame cache is
full (e.g., the encoder taking too long to return frames), it will not
cache a new frame, but instead will just increment the counter on the
last frame in the cache to schedule that frame to encode again, ensuring
that frames are on time and reducing CPU usage by lowering video
complexity. If the graphics thread takes too long to render a frame,
then it will add that frame with the count value set to the total amount
of frames that were missed (actual legitimately duplicated frames).
Because the cache gives many frames of breathing room for the encoder to
encode frames, this design helps improve results especially when using
encoding presets that have higher complexity and CPU usage, minimizing
the risk of needlessly skipped or duplicated frames.
I also managed to sneak in what should be a bit of an optimization to
reduce copying of frame data, though how much of an optimization it
ultimately ends up being is debatable.
So to sum it up, this commit increases accuracy of frame timing,
completely removes stray frame duplication, gives better results for
higher complexity encoding presets, and potentially optimizes the frame
pipeline a tiny bit.
The boolean variables which stored whether frames have been
rendered/downloaded/converted/etc were not being reset when video
restarted, causing frames to not be sent in the correct order whenever
video was reset. This could lead to minor desync of video/audio.
The 'sent_headers' member variable of the FLV output would not be reset
when the output was restarted, causing important data to not be written,
thus creating an invalid FLV file.
In certain circumstances where the output was stopping, and where data
took a long enough time to send (such as when using an encoding preset
that causes high CPU usage), the output would sometimes still send data
even after it was stopped, typically causing the output to crash.
I unintentionally made it use obs_source::sample_info instead of using
the actual target channel count, which is designated by the OBS output
sampler info. obs_source::sample_info is actually used to indicate the
currently set sampler information for incoming samples. So if a source
is outputting 5.1 channel 48khz audio, and OBS is running at stereo
44.1khz, then the obs_source::sample_info value would be set to
5.1/48khz, not the other way around. It indicates what the source
itself is running at, not what OBS is running at.
I suppose the variable needs a better name because even I used it
incorrectly despite actually having been the one who wrote it.
This dialog gives options such as increasing audio past 100%, forcing
the audio of a source to mono, and setting the audio sync offset of a
source (which was an oft-requested feature)
The copy_audio_data function really shouldn't be inlined because it's
being called twice. It's somewhat unnecessary, I think I left it inline
by accident.
This changes the way source volume handles transitioning between being
active and inactive states.
The previous way that transitioning handled volume was that it set the
presentation volume of the source and all of its sub-sources to 0.0 if
the source was inactive, and 1.0 if active. Transition sources would
then also set the presentation volume for sub-sources to whatever their
transitioning volume was. However, the problem with this is that the
design didn't take in to account if the source or its sub-sources were
active anywhere else, so because of that it would break if that ever
happened, and I didn't realize that when I was designing it.
So instead, this completely overhauls the design of handling
transitioning volume. Each frame, it'll go through all sources and
check whether they're active or inactive and set the base volume
accordingly. If transitions are currently active, it will actually walk
the active source tree and check whether the source is in a
transitioning state somewhere.
- If the source is a sub-source of a transition, and it's not active
outside of the transition, then the transition will control the
volume of the source.
- If the source is a sub-source of a transition, but it's also active
outside of the transition, it'll defer to whichever is louder.
This also adds a new callback to the obs_source_info structure for
transition sources, get_transition_volume, which is called to get the
transitioning volume of a sub-source.
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.
jp9000
fryshorts
David McMackins II