Add a new algorithm to calculate the true-peak. It implements the
Whittaker- Shannon interpolation from four samples to create 4
intermediate samples (5 x oversampling) inbetween the middle two
samples.

With 4 samples and 4 intermediate samples the algorithm can be
implemented as a 4x4 vector-matrix cross product, which is ideal for
SSE.

I've also replaced the sample-peak algorithm using SSE as well to
improve performance.

Closes obsproject/obs-studio#1189

The following features have been added to the audio-meters:
 * Stereo PPM-level meter, with 40 dB/1.7s decay rate.
 * Stereo VU-level meter, with 300 ms integration time.
 * Stereo Peak-hold meter, with 20 second sustain.
 * Input peak level color-squares in front of every meter.
 * Minor-ticks for each dB.
 * Major-ticks for every 5 dB.
 * Meter is divided in sections at -20 dB and -9 dB.

The ballistic parameters chosen here where taken from:
 * https://en.wikipedia.org/wiki/Peak_programme_meter (SMPTE RP.0155)
 * https://en.wikipedia.org/wiki/VU_meter

In the rework I have removed any ballistic calculations from
libobs/obs-audio-controls.c making the calculations here a lot more
simple doing only MAX and RMS calculations for only the samples in
the current update. The actual ballistics are now done by just
the UI/volume-control.cpp because ballistics need to be updated
based on the repaint-rate of the user-interface.

The dB to pixel conversion has been moved from
libobs/obs-audio-controls.c to UI/volume-control.cpp as well to reduce
coupling between these two objects, especially when implementing the
major- and minor-ticks and the sections.

All colors and ballistic parameters are adjustable via QT style sheets.
There are slight differences in colors for each of the themes.

Fixes an issue where the audio meter/fader would call an obs function
and lock another mutex, potentially causing a mutual inverted lock in
another thread.

(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.

Core API functions changed:
-----------------------------
EXPORT bool obs_reset_audio(struct audio_output_info *aoi);
EXPORT bool obs_get_audio_info(struct audio_output_info *aoi);

To:
-----------------------------
EXPORT bool obs_reset_audio(const struct obs_audio_info *oai);
EXPORT bool obs_get_audio_info(struct obs_audio_info *oai);

Core structure added:
-----------------------------
struct obs_audio_info {
	uint32_t            samples_per_sec;
	enum speaker_layout speakers;
	uint64_t            buffer_ms;
};

Non-interleaved (planar) floating point output is standard with audio
filtering, so to prevent audio filters from having to worry about
different audio format implementations and for the sake consistency
between user interfaces, make it so that audio is always set to
non-interleaved floating point output.

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.

Signal updated volume levels when they become available in the volume
meter. The frequency of the updates can be adjusted by setting a
different update interval.

Remove the the signal handler for the volume_level signal of audio
sources from the volume meter in anticipation of using the levels
calculated in the volume meter itself.

Use the new audio_data signal from sources to calculate the volume
levels in the volume meter.

Add a property to the volume meter that specifies the time for which
peak value should be held until it is reset.

Add a property to the volume meter that specifies the length of the
interval in which the audio data should be sampled before the
audio_levels signal is emitted.

This adds the conversion function used by the old obs based on a
logarithmic scale.

This adds a new fader type that implements a position/dB mapping
as specified in IEC 60-268-18.

This adds a volume meter object to libobs that can be used by the GUI
or plugins to convert the raw audio level data from sources to values
that can easily be used to display the audio data.
The volume meter object will use the same mapping functions as the
fader object to map dB levels to a scale.

In older versions of visual studio 2013 microsoft's WORTHLESS C compiler
has a bug where it will, almost at random, not be able to handle having
variables declared in the middle of a function and give the warning:
"illegal use of this type as an expression".  It was fixed in recent
VS2013 updates, but I'm not about to force everyone to update to it.

The macro INFINITY apparently will trigger a warning with microsoft's
compiler despite being a C standard library macro.

This adds a new library of audio control functions mainly for the use in
GUIS. For now it includes an implementation of a software fader that can
be attached to sources in order to easily control the volume.
The fader can translate between fader-position, volume in dB and
multiplier with a configurable mapping function.
Currently only a cubic mapping (mul = fader_pos ^ 3) is included, but
different mappings can easily be added.

Due to libobs saving/restoring the source volume from the multiplier,
the volume levels for existing source will stay the same, and live
changing of the mapping will work without changing the source volume.