This has a couple behavioral changes. First and biggest is that querying
AL_BUFFERS_PROCESSED from a source will always return all buffers processed
when in an AL_STOPPED state. Previously all buffers would be set as processed
when first becoming stopped, but newly queued buffers would *not* be indicated
as processed. That old behavior was not compliant with the spec, which
unequivocally states "On a source in the AL_STOPPED state, all buffers are
processed."
Secondly, querying AL_BUFFER on an AL_STREAMING source will now always return
0. Previously it would return the current "active" buffer in the queue, but
there's no basis for that in the spec.
This avoids using seq_cst for loading the source state when either inside the
mixer, or otherwise protected from inconsistencies with async updates. It also
fixes potential race conditions with getting the source offset just as a source
stops.
The voices are still all allocated in one chunk to avoid memory fragmentation.
But they're accessed as an array of pointers since the size isn't static.
ALsourceProps' Send[] array is placed at the end of the struct, and given an
indeterminate size. Extra space is allocated at the end of each struct given
the number of auxiliary sends set for the device.
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.
Sometimes the mixer is temporarily prevented from applying updates, when
multiple sources need to be updated simultaneously for example, but does not
prevent mixing. If the mixer runs during that time and a voice was just
started, it would've mixed the voice without any internal properties being set
for it.
This will also allow backends to better synchronize the tracked clock time with
the device output latency, without necessarily needing to lock if the backend
API can allow for it.
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.
Unfortunately they conflict with AL_EXT_SOURCE_RADIUS, as AL_SOURCE_RADIUS and
AL_BYTE_RW_OFFSETS_SOFT share the same source property value. A replacement for
AL_SOFT_buffer_samples will eventually be made.
