When starting a voice, the source ID was set before its first update struct was
provided, creating a small window where a listener or effect slot update could
force a voice to update without it having any valid properties to update with.
Supplying the update struct first would create a different race, where the
mixer could see a voice without a source but with an update struct, causing the
update struct to be 'freed' without being applied.
The fix here is to provide the update struct before setting the source ID, and
change the mixer to ignore update structs for voices without a source ID. This
can pseudo-orphan the updates that get set on a voice just as it stops, leaving
the struct unusable until the voice is used again, or the voice gets deleted
which will clear it. But it allows the update struct to stay in place and get
applied once the voice gets a source ID.
This allows growing the array atomically with the mixer since the ALvoice
objects themselves don't move, and a new larger array of them can be swapped in
without blocking the mixer.
The padding must be constant and sample type aligned (e.g. some fixed multiple
of two bytes between the start of two consecutive frames for 16-bit output).
The intent is to always have the ability for stereo output with WASAPI even if
the device has some other unsupported configuration, as long as front-left and
front-right exist.
This simply omits the scale factor from the filter, similar to how up-sampling
does. The consequence of this is less smooth transitions when ramping the
pitch while down-sampling, but otherwise behaves fine.
This takes advantage of the fact than when increment <= 1 (when not down-
sampling), the scale factor is always 0. As a result, the scale and scale-phase
deltas never contribute to the filtered output. Removing those multiply+add
operations cuts half of the work done by the inner loop.
Sounds that do need to down-sample (when played with a high pitch, or is 48khz
on 44.1khz output, for example), still go through the normal bsinc process.
