Unfortunately PulseAudio has a habit of limiting tlength, and trying to
calculate the device's buffer length to write regardless of tlength could
result in some amount always being writable.
This needs to be handled more automatically by the mixer to work correctly.
Otherwise, requiring a property update on resume can put the source into a
playing state with the mixer never playing it, due to not having valid mixing
parameters and the mixing parameters not getting calculated because no updates
are specified by the app (and forcing an update can break deferred updates).
Similar to the recent chorus and flanger changes, the modulation delay now
swings between -n to +n, where n is less than the delay length. This brings up
a slight issue with the linear interpolation, as modff doesn't produce the
correct fraction value for interpolation (it's inverted, with 0 being closer to
the next sample and 1 being closer to the base). So it's using nearest
interpolation for now.
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.
The effects' specified delay is the average delay time, meaning the delay
offset should move between -n and +n relative to the delay, where n <= delay.
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.
Some apps don't allow selecting an audio device, and due to problems with KDE,
PulseAudio isn't allowed to move the stream after being created by default.
Each 4 related all-passes now share a structure with one delay line, which uses
an interleaved sample history. Also fixes some potential rounding problems for
delay lines with interleaved samples.
This seems to be more in-line with the intended behavior, to allow build-up and
overlap within the reverb decay, rather than a pitch-shift on input.
Unfortunately there's no readily available implementation of this reverb model
that includes modulation to compare with, so a low depth coefficient is used to
keep it very subtle.
