For playback, increment the ring buffer's write pointer before queueing audio,
to handle cases where the callback is invoked, advancing the read pointer,
before the write pointer is advanced.
For capture, limit the number of re-queued chunks to the number of fully read
chunks.
In particular, the source sample position was reduced by the size of the
next buffer list item when one is completed, rather than the size of the
one it just completed.
It's not an issue for the final mix, but if one loop has an unaligned count,
the next loop will have unaligned input and output buffer targets which can
crash the SSE mixers.
Now full right and left is +-60 degrees instead of +-30. This should help
create a smoother panning for a sound moving in front for plain stereo output
(surround sound and HRTF are not changed).
Multi-channel sources are also not affected by this change. The stereo channel
defaults of +-pi/6 (30 degrees) still correspond to full left/right panning.
This is an unfortuante discrepancy, but is necessary for AL_EXT_STEREO_ANGLES
to work.
This also uses gain stepping for changes in density or decay, so that the
late reverb tap fades smoothly between delay offsets and density gain levels
simultaneously. Now with these changes, it's preferrable to apply density gain
adjustments on late reverb input instead of output so that samples currently in
the feedback loop won't see a sudden increase or decrease in amplitude.
A similar change can probably be made for the early reflection delays to
further smooth out delay changes.
With the vector all-pass applied in a self-contained function, the individual
steps of the early and late reverb stages can be better optimized with tighter
loops. That allows for more data to be held local, resulting in less thrashing
from reloading the same values multiple times.
There is room for further improvement, depending on the length of the early
delay lines and all-pass delay lines allowing for bulk reads.
The late reverb line lengths are long enough to ensure a single process loop
won't rely on reading samples it wrote in the same call. So we can safely read
in all samples we need from the feedback buffer up front, then more efficiently
filter them.
Instead of generating both the early and late reverb samples first, then mixing
them both to output, this now generates and mixes the early reflections then
generates and mixes the late reverb. There's no reason to hold both at the same
time so this reduces the amount of temporary storage needed.
Otherwise, using the device's maximum frequency will cause the weighting
factors to shift for different sample rates, irrespective of the content being
processed. 20khz is the maximum allowed reference frequency, and also acts as
the upper limit of human hearing.
Because density/late line length changes start affecting late reverb output
right away, with samples that are still going through feedback decay and not
just new input samples, it makes more sense to correct for it on output instead
of input. This has the additional benefit of working with the output mixer's
gain fading, avoiding discontinuities from significant density gain changes.
