This improves the transition width, allowing more of the higher frequencies
remain audible. It would be preferrable to have an upper limit of 32 points
instead of 48, to reduce the overall table size and the CPU cost for down-
sampling.
Rather than storing individual pointers to filter, scale delta, phase delta,
and scale phase delta entries, per phase index, the new table layout makes it
trivial to access the per-phase filter and delta entries given the base offset
and coefficient count.
The old layout separated filters, scale deltas, phase deltas, and scale phase
deltas into separate segments that each contained a numbers of scale and phase
entries, Since processing a sample needed a filter and one of each delta entry
relating to a particular scale and phase, the memory needed would be spread
across the whole table. And since subsequent samples would use a different
phase, it would jump around the table a whole lot as well.
The new layout packs the data in a way more consistent with its use. The
filters, scale deltas, phase deltas, and scale phase deltas are interleaved,
such that for a particular scale and phase, the filter and delta entries used
are contiguous. And the phase entries for a particular scale are kept together,
so the ~500 to ~1000 samples processed per source update stay within the same
3KB to 6KB area of the 70+KB table, which is much more cache friendly.
This improves a stereo (front-left + front-right) sound "image" by generating a
front-center channel signal. Done correctly, it helps reduce the comb effects
and phase errors associated with using only two speakers to simulate center
sounds.
Note that it shouldn't be used if the front-center channel is already included
in the positional audio mix (the dialog effect is okay). In general, it may
actually be better to exclude the front-center channel from the positional
audio mix and use this to generate front-center output.
The HF absorption is applied given the source distance, as relative to the
source's immediate environment, with additional absorption being applied given
the room/reverb environment. This does double up the amount of absorption
compared to the dry path, but it can be assumed the initial reflections travel
a longer distance.
This is just for the output limiter right now, but in the future can be used
for the compressor EFX effect. The parameters are also hardcoded, but can be
made configurable after 1.18.
This properly accounts for the room rolloff factor for normal air absorption
(which makes it none by default, like distance attenuation), and uses the
reverb's decay time, decay hf ratio, decay hf limit, and room air absorption
properties to calculate an initial hf decay with the WetGainAuto flag. This
mirrors the behavior of the initial distance decay.
The previous value couldn't actually be expressed as a float and got rounded up
to the next whole number value, leaving the potential for an overrun in the
squared sum.
This reduces the output volume when the mixed samples extend outside of -1,+1,
to prevent excessive clipping. It can reduce the volume by -80dB in 50ms, and
increase it by +80dB in 1s (it will not go below -80dB or above 0dB).
