Had the audio restart slot connected to things that didn't require a
restart

The strings didn't have ending double quotes.  No clue why this didn't
fail in GCC and VC.  Well, VC is horrible but I expected better out of
GCC.

 - Fix a bug where the initial audio data insertion would cause all
   audio data to unintentionally clear (mixed up < and > operators, damn
   human error)

 - Fixed a potential interdependant lock scenario with channel mutex
   locks and graphics mutex locks.  The main video thread could lock the
   graphics mutex and then while in the graphics mutex could lock the
   channels mutex.  Meanwhile in another thread, the channel mutex could
   get locked, and then the graphics mutex would get locked, causing a
   deadlock.

   The best way to deal with this is to not let mutexes lock within
   other mutexes, but sometimes it's difficult to avoid such as in the
   main video thread.

 - Audio devices should now be functional, and the devices in the audio
   settings can now be changed as desired.

There shouldn't be errors if the actual source has successfully been
created, just warnings.

Also, revamp the settings dialog code and make it use signals and slots
a bit better.

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.

The function would fail to compile on linux because the variables didn't
exist

It will now load up a the list of audio input/output devices in the
combo boxes in audio settings.

 - 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.

I can't believe I did !res there.  Well I suppose I can believe it, but
still.

The enumeration code being up at the top was making things quite messy,
so I split that code out to a separate set of files.

 - 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

Somehow this code didn't get included with the last commit.

Also, don't have it repeat trying to reconnect if no devices are found

 - 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.

As usual, microsoft treats all enums as integers, rather than actually
even checking the enum type.  Worthless compiler.  Just complete
garbage.

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.

Happened because I compiled on windows and it only compiled
windows-specific code.

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.

Also, make it use 'veryfast' preset.  Still testing this, might have to
revise this later.

The mutex must be unlocked regardless of what the return value is or it
will be locked forever if the return value isn't 0.

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.

Forgot to add a '!'.

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.