Modify the obs_display API so that it always uses an orthographic
projection that is the size of the display, rather than OBS' base size.
Having it do an orthographic projection to OBS' base size was silly
because it meant that everything would be skewed if you wanted to draw
1:1 in the display. This deoes mean that the callbacks must handle
resizing the images, but it's worth it to ensure 1:1 draw sizes.
As for the preview widget, instead of making some funky widget within
widget that resizes, it's just going to be a widget within the entire
top layout. Also changed the preview padding color to gray.
- Implement a means of obtaining default settings for an
input/output/encoder. obs_source_defaults for example will return
the default settings for a particular source type.
- Because C++ doesn't have designated initializers, use functions in
the WASAPI plugin to register the sources instead.
Having everything in global.ini meant that if you wanted different
settings for studio mode, that it would also overwrite it for basic
mode. This way, the settings for each mode are separate, and you can
use different settings for each mode.
- Implement windows monitor capture (code is so much cleaner than in
OBS1). Will implement duplication capture later
- Add GDI texture support to d3d11 graphics library
- Fix precision issue with sleep timing, you have to call
timeBeginPeriod otherwise windows sleep will be totally erratic.
- Add WASAPI audio capture for windows, input and output
- Check for null pointer in os_dlopen
- Add exception-safe 'WinHandle' and 'CoTaskMemPtr' helper classes that
will automatically call CloseHandle on handles and call CoTaskMemFree
on certain types of memory returned from windows functions
- Changed the wide <-> MBS/UTF8 conversion functions so that you use
buffers (like these functions are *supposed* to behave), and changed
the ones that allocate to a different naming scheme to be safe
- Split input and output audio captures so that they're different
sources. This allows easier handling and enumeration of audio
devices without having to do some sort of string processing.
This way the user interface code can handle this a bit more easily,
and so that it doesn't confuse users either. This should be done for
all audio capture sources for all operating systems. You don't have
to duplicate any code, you just need to create input/output wrapper
functions to designate the audio as input or output before creation.
- Make it detect soundflower and wavtap devices as mac "output" devices
(even though they're actually input) for the mac output capture, and
make it so that users can select a default output capture and
automatically use soundflower or wavtap.
I'm not entirely happy about having to do this, but because mac is
designed this way, this is really the only way to handle it that
makes it easier for users and UI code to deal with.
Note that soundflower and wavtap are still also designated as input
devices, so will still show up in input device enumeration.
- Remove pragma messages because they were kind polluting the other
compiler messages and just getting in the way. In the future we can
just do a grep for TODO to find them.
- Redo list property again, this time using a safer internal array,
rather than requiring sketchy array inputs. Having functions handle
everything behind the scenes is much safer.
- Remove the reference counter debug log code, as it was included
unintentionally in a commit.
Categories added an unnecessary complexity to making properties, and
would very likely almost never be used in most cases, and were more of a
display feature. The main issue is that it made property data more
complex to work with, and I just didn't feel comfortable with that.
Also, added a function to allow you to retrieve a porperty just by its
name.
When a source/output/etc has a property of a 'list' type, there was no
way to get the names associated with its values. That, and it only
supported lists of either text, or enums (0..[value] only).
Now, you can associate translated names with those values, and use
integer, float, or string values. Put it all in to one function as well
to simplify its usage.
I plan on using this to help get enumerations from devices/etc for
certain types of sources. For example, if I get the properties of an
audio source, I'd like to have a list of available devices with it as
well.
- Signals and dynamic callbacks now require declarations to be made
before being used. What this does is allows us to get information
about the functions dynamically which can be relayed to the user and
plugins for future extended usage (this should have big implications
later for scripting in particular, hopefully).
- Reduced the number of types calldata uses from "everything I could
think of" to simply integer, float, bool, pointer/object, string.
Integer data is now stored as long long. Floats are now stored as
doubles (check em).
- Use a more consistent naming scheme for lexer error/warning macros.
- Fixed a rather nasty bug where switching to an existing scene would
cause it to increment sourceSceneRefs, which would mean that it would
never end up never properly removing the source when the user clicks
removed (stayed in limbo, obs_source_remove never got called)
If the default device changes, set the reconnect interval to 200
milliseconds so it pretty much immediately tries to reinitialize the
audio with the newly selected default device. Otherwise, use 2000
millisecond intervals, and assume disconnection.
Also, reduced FFmpeg logging to just regular FFmpeg information rather
than everything FFmpeg logs.
See, it can sometimes be a bit confusing. These functions should
definitely not fail under normal circumstances, and these errors may
affect the user and/or application in some way.
LOG_ERROR should be used in places where though recoverable (or at least
something that can be handled safely), was unexpected, and may affect
the user/application.
LOG_WARNING should be used in places where it's not entirely unexpected,
is recoverable, and doesn't really affect the user/application.
I can't believe I wasn't doing this. This is why file output was
getting corrupted. Audio and video send in data from separate threads.
I should be embarassed for not having considered that.
Key lesson: Increase threading paranoia levels. Apparently my
threading paranoid levels are lackluster.
I'm not entirely sure what's going on with my FFmpeg code, but it's
definitely not generating the proper footers for MP4 files, despite the
fact that the footer function succeeds. Going to use AVIs for the time
being still.
After a mac just boots up, it often takes about 700 milliseconds for
audio devices to work on first use, so it would often have issues with
the 700ms audio buffering time, and audio data would get cut off. Just
increasing the buffering a little bit fixes the issue.
Actually, if initializing failed at all, it would never properly
uninitialize because the 'initialized' variable was never set until the
very end. Instead, set the "initialized" flag from the beginning to
ensure initialization.
- Add CoreAudio device input capture for mac audio capturing. The code
should cover just about everything for capturing mac input device
audio. Because of the way mac audio is designed, users may have no
choice but to obtain the open source soundflower software to capture
their mac's desktop audio. It may be necessary for us to distribute
it with the program as well.
- Hide event backend
- Use win32 events for windows
- Allow timed waits for events
- Fix a few warnings
FFmpeg test output wasn't make any attempt to sync data before. Should
be much more accurate now.
Also, added a restart message to audio settings if base audio settings
are changed.
the signals for scenes could have potentially conflicted with default
source signals. "remove" should be used for source removal, for
example. Changed the scene signals to "item-add" and "item-remove" for
its items.
Resetting audio while libobs is active is a real pain. I think I'm just
going to do audio resetting later, or maybe just require restart
regardless just because having to shut down audio streams/lines while
there's sources currently active requires recreating all the audio
lines for each audio source. Very painful.
Video fortunately is no big deal, so at least there's that.
Split off activate to activate and show callbacks, and split off
deactivate to deactivate and hide callbacks. Sources didn't previously
have a means to know whether it was actually being displayed in the main
view or just happened to be visible somewhere. Now, for things like
transition sources, they have a means of knowing when they have actually
been "activated" so they can initiate their sequence.
A source is now only considered "active" when it's being displayed by
the main view. When a source is shown in the main view, the activate
callback/signal is triggered. When it's no longer being displayed by
the main view, deactivate callback/signal is triggered.
When a source is just generally visible to see by any view, the show
callback/signal is triggered. If it's no longer visible by any views,
then the hide callback/signal is triggered.
Presentation volume will now only be active when a source is active in
the main view rather than also in auxilary views.
Also fix a potential bug where parents wouldn't properly increment or
decrement all the activation references of a child source when a child
was added or removed.
Implement a few audio options in to the user interface as well as a few
inline audio functions in audio-io.h.
Make it so ffmpeg plugin automatically converts to the desired format.
Use regular interleaved float internally for audio instead of planar
float.
jp9000