Rather than each buffer being individually allocated with a generated 'thunk'
ID that's used with a uint:ptr map, buffers are allocated in arrays of 64
within a vector. Each group of 64 has an associated 64-bit mask indicating
which are free to use, and the buffer ID is comprised of the two array indices
which directly locate the buffer (no searching, binary or otherwise).
Currently no buffers are actually deallocated after being allocated, though
they are reused. So an app that creates a ton of buffers once, then deletes
them all and uses only a couple from then on, will have a bit of waste, while
an app that's more consistent with the number of used buffers won't be a
problem. This can be improved by removing elements of the containing vector
that contain all-free buffers while there are plenty of other free buffers.
Also, this method can easily be applied to other resources, like sources.
Requires the MAP_READ_BIT or MAP_WRITE_BIT flags to be OR'd with the format
upon a call to alBufferData, to enable mappable storage for the given access
types. This will fail if the format requires internal conversion and doesn't
resemble the original input data, so the app can be guaranteed the size, type,
and layout of the original data is the same as what's in storage.
Then alMapBufferSOFT may be called with appropriate bit flags to get a readable
and/or writable pointer to the buffer's sample storage. alUnmapBufferSOFT must
be called when access is finished. It is currently invalid to map a buffer that
is attached to a source, or to attach a buffer to a source that is currently
mapped. This restriction may be eased in the future, at least to allow read-
only access while in use (perhaps also to allow writing, if coherency can be
achieved).
Currently the access flags occupy the upper 8 bits of a 32-bit bitfield to
avoid clashing with format enum values, which don't use more than 16 or 17
bits. This means any future formats are limited to 24-bit enum values, and also
means only 8 flags are possible when declaring storage. The alternative would
be to add a new function (alBufferStorage?) with a separate flags parameter.
This will be to allow buffer layering, multiple buffers of the same format and
sample rate that are mixed together prior to resampling, filtering, and
panning. This will allow composing sounds from individual components that can
be swapped around on different invocations (e.g. layer SoundA and SoundB on one
instance and SoundA and SoundC on a different instance for a slightly different
sound, then just SoundA for a third instance, and so on). The longest buffer
within the list item determines the length of the list item.
More work needs to be done to fully support it, namely the ability to specity
multiple buffers to layer for static and streaming sources. Also the behavior
of loop points for layered static sources should be worked out. Should also
consider allowing each layer to have a sample offset.
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).
Clang does not allow using C11's atomic_load on const _Atomic variables.
Previously it just disabled use of C11 atomics if atomic_load didn't work on a
const _Atomic variable, but I think I'd prefer to have Clang use C11 atomics
for the added features (more explicit memory ordering) even if it means a few
instances of breaking const.
NFC filters currently only work when rendering to ambisonic buffers, which
includes HQ rendering and ambisonic output. There are two new config options:
'decoder/nfc' (default on) enables or disables use of NFC filters globally, and
'decoder/nfc-ref-delay' (default 0) specifies the reference delay parameter for
NFC-HOA rendering with ambisonic output (a value of 0 disables NFC).
Currently, NFC filters rely on having an appropriate value set for
AL_METERS_PER_UNIT to get the correct scaling. HQ rendering uses the averaged
speaker distances as a control/reference, and currently doesn't correct for
individual speaker distances (if the speakers are all equidistant, this is
fine, otherwise per-speaker correction should be done as well).
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.
