The source's voice holds a copy of the last properties it received, so listener
updates can make sources recalculate internal properties from that stored copy.
This necessitates a change in how source updates are handled. Rather than just
being able to update sources when a dependent object state is changed (e.g. a
listener gain change), now all source updates must be proactively provided.
Consequently, apps that do not utilize any deferring (AL_SOFT_defer_updates or
alcSuspendContext/alcProcessContext) may utilize more CPU since it'll be
filling out more update containers for the mixer thread to use.
The upside is that there's less blocking between the app's calling thread and
the mixer thread, particularly for vectors and other multi-value properties
(filters and sends). Deferring behavior when used is also improved, since
updates that shouldn't be applied yet are simply not provided. And when they
are provided, the mixer doesn't have to ignore them, meaning the actual
deferring of a context doesn't have to synchrnously force an update -- the
process call will send any pending updates, which the mixer will apply even if
another deferral occurs before the mixer runs, because it'll still be there
waiting on the next mixer invocation.
There is one slight bug introduced by this commit. When a listener change is
made, or changes to multiple sources while updates are being deferred, it is
possible for the mixer to run while the sources are prepping their updates,
causing some of the source updates to be seen before the other. This will be
fixed in short order.
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.
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.
At 0 distance from the listener, the sound is omni-directional. As the source
and listener become 'radius' units apart, the sound becomes more directional.
With HRTF, an omni-directional sound is handled using 0-delay, pass-through
filter coefficients, which is blended with the real delay and coefficients as
needed to become more directional.
