It still fades between HRIRs when it changes, but now it selects the nearest
one instead of blending the nearest four. Due to the minimum-phase nature of
the HRIRs, interpolating between delays lead to some oddities which are
exasperated by the fading (and the fading is needed to avoid clicks and pops,
and smooth out changes).
Technically it uses A-Format processing from the B-Format input and output. But
this attempts to provide better spatial definition to the reverberation so that
it can be used in a more generic fashion, allowing it to be decoded as any
other B-Format signal to whatever output is needed, and also allowing for a bit
of height information when the output is capable of such.
There may still be some kinks to work out, such as properly decorrelating the
early reflection taps and tweaking the late reverb density. But it seems to be
a good enough start.
This uses an AllRAD-derived decoder matrix for the high frequencies, which
seems to improve positioning response. It also switches back to dual-band.
The low frequencies appear to be unexpectedly quiet by comparison, but it's not
that bad and can be tweaked later.
Designed for apps that either don't change the listener's AL_GAIN, or don't
allow the listener's AL_GAIN to go above 1. This allows the volume to still be
increased further than such apps may allow, if users find it too quiet.
Be aware that increasing this can easily cause clipping. The gain limit
reported by AL_GAIN_LIMIT_SOFT is also affected by this.
It's the only implementation currently, so there's no point to having it stored
as a function pointer in the filter struct. Even if there were SIMD versions,
it'd be a global selection, not per-instance.
Since it was merely acting as an extension of it anyway, with the second delay
line tap (for late reverb) copying attenuated samples to the decorrelator line
that was being tapped off of. Just extend the delay line and offset the
decorrelator taps to be relative to the late reverb tap.
Since it's accumulating multiple HRIRs for two output speakers, it seems to be
a better option to preserve the amplitude of the high-frequency decoder instead
of increasing it, and reduce the amplitude of the low-frequency decoder to
compensate.
14 in total, an 8-point cube and a 6-point diamond shape, to help improve sound
localization a bit. Incurs no real extra CPU cost once the IRs are built.
Less than ideal since documentations warn it may not list 'neon' even if it's
really supported. However, the "proper" APIs to check for NEON extensions don't
seem to exist in my toolchain.
Ideally the band-pass should probably happen closer to output, like gain is.
However, doing that would require 16 filters (4 early + 4 late channels, each
with a low-pass and high-pass filter), compared to the two needed to do it on
input.
The combined source and listener gains now can't exceed a multiplier of 16
(~24dB). This is to avoid mixes getting out of control with large volume
boosts, which reduces the effective precision given by floating-point.
This appears to be how Creative's Windows drivers handle it, and is necessary
for at least the Windows version of UT2k4 (otherwise it tries to play a source
while suspended, checks and sees it's stopped, then kills it before it's given
a chance to start playing).
Consequently, the internal properties it gets mixed with are determined by what
the source properties are at the time of the play call, and the listener
properties at the time of the suspend call.
This does not change alDeferUpdatesSOFT, which will still hold the play state
change until alProcessUpdatesSOFT.
