The event functions got renamed in obs.

Didn't convert the event names and didn't have the right mac includes

...The reason why audio didn't work was because I overwrote the bitrate
values.

As for semaphores, mac doesn't support unnamed semaphores without using
mach semaphores.  So, I just implemented a semaphore wrapper for each
OS.

Also, check that audio devices are available before setting defaults

 - Add some temporary streaming code using FFmpeg.  FFmpeg itself is not
   very ideal for streaming; lack of direct control of the sockets and
   no framedrop handling means that FFmpeg is definitely not something
   you want to use without wrapper code.  I'd prefer writing my own
   network framework in this particular case just because you give away
   so much control of the network interface.  Wasted an entire day
   trying to go through FFmpeg issues.

   There's just no way FFmpeg should be used for real streaming (at
   least without being patched or submitting some sort of patch, but I'm
   sort of feeling "meh" on that idea)

   I had to end up writing multiple threads just to handle both
   connecting and writing, because av_interleaved_write_frame blocks
   every call, stalling the main encoder thread, and thus also stalling
   draw signals.

 - Add some temporary user interface for streaming settings.  This is
   just temporary for the time being.  It's in the outputs section of
   the basic-mode settings

 - Make it so that dynamic arrays do not free all their data when the
   size just happens to be reduced to 0.  This prevents constant
   reallocation when an array keeps going from 1 item to 0 items.  Also,
   it was bad to become dependent upon that functionality.  You must now
   always explicitly call "free" on it to ensure the data is free, and
   that's how it should be.  Implicit functionality can lead to
   confusion and maintainability issues.

 - 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.

 - 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.

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.

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.

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.

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

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.

If coreaudio_start failed, it wouldn't free the audio data properly.
Fixed that issue.

These address structures are very confusing and I wish apple designed
better system APIs.

 - 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.

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.

Add a scaler interface (defaults to swscale), and if a separate output
wants to use a different scale or format than the default output format,
allow a scaler instance to be created automatically for that output,
which will then receive the new scaled output.

av_frame_alloc, av_frame_free, among others, live in libavutil

If there are for example more than one audio outputs and they have
different sample rates or channels and such, this will allow automatic
conversion of that audio to the request formats/channels/rates (but only
if requested).

 - Changed glMapBuffer to glMapBufferRange to allow invalidation.  Using
   just glMapBuffer alone was causing some unacceptable stalls.

 - Changed dynamic buffers from GL_DYNAMIC_WRITE to GL_STREAM_WRITE
   because I had misunderstood the OpenGL specification

 - Added _OPENGL and _D3D11 builtin preprocessor macros to effects to
   allow special processing if needed

 - Added fmod support to shaders (NOTE: D3D and GL do not function
   identically with negative numbers when using this.  Positive numbers
   however function identically)

 - Created a planar conversion shader that converts from packed YUV to
   planar 420 right on the GPU without any CPU processing.  Reduces
   required GPU download size to approximately 37.5% of its normal rate
   as well.  GPU usage down by 10 entire percentage points despite the
   extra required pass.

There were a *lot* of warnings, managed to remove most of them.

Also, put warning flags before C_FLAGS and CXX_FLAGS, rather than after,
as -Wall -Wextra was overwriting flags that came before it.

The API used to be designed in such a way to where it would expect
exports for each individual source/output/encoder/etc.  You would export
functions for each and it would automatically load those functions based
on a specific naming scheme from the module.

The idea behind this was that I wanted to limit the usage of structures
in the API so only functions could be used.  It was an interesting idea
in theory, but this idea turned out to be flawed in a number of ways:

 1.) Requiring exports to create sources/outputs/encoders/etc meant that
     you could not create them by any other means, which meant that
     things like faruton's .net plugin would become difficult.

 2.) Export function declarations could not be checked, therefore if you
     created a function with the wrong parameters and parameter types,
     the compiler wouldn't know how to check for that.

 3.) Required overly complex load functions in libobs just to handle it.
     It makes much more sense to just have a load function that you call
     manually.  Complexity is the bane of all good programs.

 4.) It required that you have functions of specific names, which looked
     and felt somewhat unsightly.

So, to fix these issues, I replaced it with a more commonly used API
scheme, seen commonly in places like kernels and typical C libraries
with abstraction.  You simply create a structure that contains the
callback definitions, and you pass it to a function to register that
definition (such as obs_register_source), which you call in the
obs_module_load of the module.

It will also automatically check the structure size and ensure that it
only loads the required values if the structure happened to add new
values in an API change.

The "main" source file for each module must include obs-module.h, and
must use OBS_DECLARE_MODULE() within that source file.

Also, started writing some doxygen documentation in to the main library
headers.  Will add more detailed documentation as I go.

It will now output to a a filename that you specify when you click the
record button.  This is just for testing.