This appears to be how Creative's Windows drivers handle it, and is necessary
for at least the Windows version of UT2k4 (otherwise it tries to play a source
while suspended, checks and sees it's stopped, then kills it before it's given
a chance to start playing).
Consequently, the internal properties it gets mixed with are determined by what
the source properties are at the time of the play call, and the listener
properties at the time of the suspend call.
This does not change alDeferUpdatesSOFT, which will still hold the play state
change until alProcessUpdatesSOFT.
Last time this attempted to average the HRIRs according to their contribution
to a given B-Format channel as if they were loudspeakers, as well as averaging
the HRIR delays. The latter part resulted in the loss of the ITD (inter-aural
time delay), a key component of HRTF.
This time, the HRIRs are averaged similar to above, except instead of averaging
the delays, they're applied to the resulting coefficients (for example, a delay
of 8 would apply the HRIR starting at the 8th sample of the target HRIR). This
does roughly double the IR length, as the largest delay is about 35 samples
while the filter is normally 32 samples. However, this is still smaller the
original data set IR (which was 256 samples), it also only needs to be applied
to 4 channels for first-order ambisonics, rather than the 8-channel cube. So
it's doing twice as much work per sample, but only working on half the number
of samples.
Additionally, since the resulting HRIRs no longer rely on an extra delay line,
a more efficient HRTF mixing function can be made that doesn't use one. Such a
function can also avoid the per-sample stepping parameters the original uses.
Currently incomplete, as second- and third-order output will not correctly
handle B-Format input buffers. A standalone up-sampler will be needed, similar
to the high-quality decoder.
Also, output is ACN ordering with SN3D normalization. A config option will
eventually be provided to change this if desired.
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 uses a separate container to provide the relevant properties to the
internal update method, using atomic pointer swaps. A free-list is used to
avoid having too many individual containers.
This allows the mixer to update the internal listener properties without
requiring the lock to protect against async updates. It also allows concurrent
read access to the user-facing property values, even the multi-value ones (e.g.
the vectors).
Instead of looping over all the coefficients for each channel with multiplies,
when we know only one will have a non-0 factor for ambisonic mixing buffers,
just index the one with a non-0 factor.
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.
