| Commit message (Collapse) | Author | Age | Files | Lines |
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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.
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i.e. without the latency
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The only mixer locking involved is with the backend, as determined by it's
ability to get the device clock and latency atomically.
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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.
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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.
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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.
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This means we track the current params and the target params, rather than the
target params and the stepping. This closer matches the non-HRTF mixers.
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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.
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The sound localization with virtual channel mixing was just too poor, so while
it's more costly to do per-source HRTF mixing, it's unavoidable if you want
good localization.
This is only partially reverted because having the virtual channel is still
beneficial, particularly with B-Format rendering and effect mixing which
otherwise skip HRTF processing. As before, the number of virtual channels can
potentially be customized, specifying more or less channels depending on the
system's needs.
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This new method mixes sources normally into a 14-channel buffer with the
channels placed all around the listener. HRTF is then applied to the channels
given their positions and written to a 2-channel buffer, which gets written out
to the device.
This method has the benefit that HRTF processing becomes more scalable. The
costly HRTF filters are applied to the 14-channel buffer after the mix is done,
turning it into a post-process with a fixed overhead. Mixing sources is done
with normal non-HRTF methods, so increasing the number of playing sources only
incurs normal mixing costs.
Another benefit is that it improves B-Format playback since the soundfield gets
mixed into speakers covering all three dimensions, which then get filtered
based on their locations.
The main downside to this is that the spatial resolution of the HRTF dataset
does not play a big role anymore. However, the hope is that with ambisonics-
based panning, the perceptual position of panned sounds will still be good. It
is also an option to increase the number of virtual channels for systems that
can handle it, or maybe even decrease it for weaker systems.
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Also only access the activesource's source field once per update.
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