Fixes a bug introduced in 6407707a where if you open the settings
window the settings would always think there were new changes.  It was
marking a control as changed when it shouldn't be

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This variable is used to detect whether audio has stopped -- if audio
stops, it detects that no new data is coming in, and resets the audio
position so that it eliminates the chance of causing the audio buffering
to go haywire if audio starts up again.  However, this variable was not
being reset every time the value changes, which it should.

Sometimes the A and B sources of a transition would a large difference
in their timestamps, and the calculation of where to start the audio
data for one of the sources could be above the tick size, which could
cause a crash.

If the circular audio buffer of the source has data remaining that's
less than the audio frame tick count (1024 frames), it would just leave
that audio data on the source without discarding it.  However, this
could cause audio buffering to increase unnecessarily under certain
circumstances (when the next audio timestamp is within the timestamp
jump window), so it would append data to that circular buffer despite
the audio stopping that long ago, causing audio buffering to have to
increase to compensate.

Instead, just discard pending audio if it hasn't been written to.  In
other words, if the audio has stopped and there's insufficient audio
left to continue processing.

With the new audio subsystem, audio buffering is minimal at all times.
However, when the audio buffering is too small or non-existent, it would
cause the audio encoders to start with a timestamp that was actually
higher than the first video frame timestamp.  Video would have some
inherent buffering/delay, but then audio could return and encode almost
immediately.  This created a possible window of empty time between the
first encoded video packet and the first encoded audio packet, where as
audio buffering would cause the first audio packet's timestamp to always
be way before the first video packet's timestamp.  It would then
incorrectly assume the two starting points were in sync.

So instead of assuming the audio data is always first, this patch makes
video wait for audio data comes in, and conversely buffers audio data
until video comes in, and tries to find a starting point within that
video data instead, ensuring a synced starting point whether audio
buffering is active or not.

When starting a multi-track output, attempt to pair the video encoder
with one of the audio encoders to ensure that the video and audio
encoders start as close together in time as possible.  This ensures the
best possible audio/video syncing point when using multi-track audio
output.

When using multi-track audio, encoders cannot be paired like they can
when only using a single audio track with video, so it has to choose the
best point in the interleaved buffer as the "starting point", and if the
encoders start up at different times, it has to prune that data and wait
to start the output on the next video keyframe.  When the audio encoders
started up, there was the case where the encoders would take some time
to load, and it would cause the pruning code to wait for the next
keyframe to ensure startup syncing.

Starting the audio encoders before starting the video encoder should
reduce the possibility of that happening in a multi-track scenario.

In a multi-track scenario it was not taking in to consideration the
possibility of secondary audio tracks, which could have caused desync on
some of the audio tracks.

Gets the index of the first interleaved packet of a given/type and audio
index

audio_input_buf should never be accessed outside of audio_buf_mutex.

The seamless audio looping code would erroneously trigger for things
that weren't loops, causing the audio data to continually push back and
ignore timestamps, thus going out of sync.

There does need to be loop handling code, but due to the fact that other
things may need to trigger this code, it's best just to clear the audio
data and start from a fresh sync point.  Unfortunately for the case of
loops, this means the window in which audio data loops and video frames
loop need to be muted.

This is an additional method/helper that prevents composite sources from
treating non-audio sources as audio sources.

Fixes an issue where audio data would not be popped if they were not
activated/presenting.  This would cause the audio subsystem to
needlessly buffer when they were reactivated again.  Rendering all audio
sources (excuding composite/filter sources) helps ensure that audio data
is always popped and not left to pile up.

When the new UI was put in, the action shortcuts were unintentionally
removed from the list boxes.

UI: Fix GCC 4.8 compilation issue

Apparently using QPointer container for the QObject being given to the
connect function makes it so it can't find the actual function to use.
I'm guessing this is incomplete functionality or a bug that existed in
GCC 4.8.  Doesn't happen in 4.9+.

The "full_range" member variable of the video frame was being passed
"false" each time even though it may not necessarily be set to partial
range

Implements transitions, and introduces "Studio Mode" which allows live
editing of the same or different scenes while preserving what's
currently being displayed.

Studio Mode offers a number of new features:
  - The ability to edit different scenes or the same scene without
    modifying what's currently being displayed (of course)

  - The ability to set up "quick transitions" with a desired transition
    and duration that can be assigned hotkeys

  - The option to create full copies of all sources in the program scene
    to allow editing of source properties of the same scene live without
    modifying the output, or (by default) just use references.  (Note
    however that certain sources cannot be duplicated, such as capture
    sources, media sources, and device sources)

  - Swap Mode (enabled by default) which swaps the program scene with
    the preview scene when a transition completes

Currently, only non-configurable transitions (transitions without
properties) are listed, and the only transitions available as of this
writing are fade and cut.  In future versions more transitions will be
added, such as swipe, stingers, and many other various sort of
transitions, and the UI will support being able to add/configure/remove
those sort of configurable transitions.

Makes signals use stack memory rather than allocate memory each time.
Most likely a completely insignificant and pointless optimization.

A comment that serves as a reminder to anyone who might need to edit the
scene code.  If the graphics mutex must be locked, it must be locked
first before entering the scene mutexes, or outside of the scene
mutexes.

This fixes an age-old issue where audio samples could be lost or audio
could temporarily go out of sync in the case of looping videos.  When
audio/video data is looping, there's a window between when the audio
data resets its timestamp value and when the video data resets its
timestamp value.  This method simply pushes back the audio data while in
that window and does not modify sync, and when it detects that its out
of the loop window it simply forces a resync of the audio data in the
circular buffer.

This ensures that minimal audio data is lost in the loop process, and
minimizes the likelihood of any sort of sync issues associated with
looping.

Apply user sync offset *after* timestamp smoothing, not before.
Prevents small or gradual sync offsets from not being properly applied.

Instead of applying the resampler offset right away (to each audio
packet), apply the resampler offset when the timestamps are converted to
system timestamps.  This fixes an issue where if audio timestamps reset
to 0 (for whatever reason), the offset would cause the timestamp to go
in to the negative.

(Note: This commit also modifies the UI)

Allows the ability to duplicate sources fully copied, and/or have the
scene and its duplicates be private sources

Creates a scene marked as a private source

Allows full duplication of sources (with exception of sources that are
marked with the OBS_SOURCE_DO_NOT_DUPLICATE output capability flag)

Certain types of sources (display captures, game captures, audio
device captures, video device captures) should not be duplicated.  This
capability flag hints that the source prefers references over full
duplication.

Mostly only used for transitions with the intention of automatically
creating transitions which don't require configuration, returns whether
the source has any properties or not (whether it's configurable)

Changes it to use obs_context_by_name

Allows creation of private/unlisted sources/outputs/services/encoders

The intention of this is to allow sources/outputs/etc to be created
without being visible to the UI or save/load functions.

(Note: This commit also modifies UI)

Instead of using signals, use designated callback lists for audio
capture and audio control helpers.  Signals aren't suitable here due to
the fact that signals aren't meant for things that happen every frame or
things that happen every time audio/video is received.  Also prevents
audio from being allocated every time these functions are called due to
the calldata structure.