Could really do with some optimizations to the mixing gain calculations. For
ambisonic targets, the coefficients will only have 1 non-0 entry for each
output, so the double loop in unnecessarily wasteful. Similarly, most uses
won't need a full height encoding either, so a horizontal-only or mixed-order
target could reduce the number of channels.
This uses a virtual B-Format buffer for mixing, and then uses a dual-band
decoder for improved positional quality. This currently only works with first-
order output since first-order input (from the AL_EXT_BFROMAT extension) would
not sound correct when fed through a second- or third-order decoder.
This also does not currently implement near-field compensation since near-field
rendering effects are not implemented.
There were phase issues caused by applying HRTF directly to the B-Format
channels, since the HRIR delays were all averaged which removed the inter-aural
time-delay, which in turn removed significant spatial information.
This mixes to a 4-channel first-order ambisonics buffer. With ACN ordering and
N3D scaling, this makes it easy to remain compatible with effects that only
care about mono input since channel 0 is an unattenuated mono signal.
This helps the stability of transforms to local space for sources that are at
or near the listener. With a single-precision matrix, even FLT_EPSILON might
not be enough to detect matching positions.
This is essentially a 12-point sinc resampler, unless it's resampling to a rate
higher than the output, at which point it will vary between 12 and 24 points
and do anti-aliasing to avoid/reduce frequencies going over nyquist.
Code provided by Christopher Fitzgerald.
The EAX-only effect properties will be set to compatible defaults when standard
reverb is set, and the EAX-only effects will be skipped during sample
processing.
If the filter properties are continually updated, and the HF or LF gain goes
from <1, to 1, and later back to <1, the history shouldn't hold stale values
from before it was at 1.
The extension's not going anywhere, and it can't do anything fluidsynth can't.
The code maintenance and bloat is not worth keeping around, and ideally the AL
API would be able to facilitate MIDI-like behavior anyway (envelopes, start-at-
time, etc).
This helps avoid different results when looping is toggled within a couple
samples of the loop point, or when a processed buffer is removed while the
source is only a couple samples into the next buffer.
Chris Robinson