| Commit message (Collapse) | Author | Age | Files | Lines |
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This avoids having to clamp the fade value when incrementing it.
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This update modifies the reverb in numerous ways. The 3-series, 4-parallel
all-pass is replaced with a Gerzon vector all-pass. The vector all-pass is also
applied to the early reflections, to help with the initial diffusion in the
late reverb. The late reverb filter and feedback attenuation has been replaced
with a dual first-order equalization filter, which controls the low and high
frequencies with individual low-pass/high-shelf and high-pass/low-shelf filters
with gain control.
Additionally, delay lines now have the ability to blend the output between two
offsets, to help smooth out the transition for when the delay changes (without
such, it could result in undesirable clicks and pops).
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This attempts to improve the smoothness of the late reverb decay by passing
each line through multiple all-pass filters. Some work is still needed to work
better in high-density and not-so-high-diffusion environments.
This also removes the decay from the early reflections, since it's no longer
continuous feedback.
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Also clean up comment formatting a bit.
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Technically it uses A-Format processing from the B-Format input and output. But
this attempts to provide better spatial definition to the reverberation so that
it can be used in a more generic fashion, allowing it to be decoded as any
other B-Format signal to whatever output is needed, and also allowing for a bit
of height information when the output is capable of such.
There may still be some kinks to work out, such as properly decorrelating the
early reflection taps and tweaking the late reverb density. But it seems to be
a good enough start.
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It's the only implementation currently, so there's no point to having it stored
as a function pointer in the filter struct. Even if there were SIMD versions,
it'd be a global selection, not per-instance.
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Since it was merely acting as an extension of it anyway, with the second delay
line tap (for late reverb) copying attenuated samples to the decorrelator line
that was being tapped off of. Just extend the delay line and offset the
decorrelator taps to be relative to the late reverb tap.
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Ideally the band-pass should probably happen closer to output, like gain is.
However, doing that would require 16 filters (4 early + 4 late channels, each
with a low-pass and high-pass filter), compared to the two needed to do it on
input.
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This is mostly just reorganizing the effects to call the Construct method which
initializes the ref count.
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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.
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It's been disabled forever, and I have no idea how to make it work properly.
Better to just redo it when making something that works.
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It's a horriobly inefficient way to process multiple samples through the
filter.
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Similar to the listener, separate containers are provided atomically for the
mixer thread to apply updates without needing to block, and a free-list is used
to reuse container objects.
A couple things to note. First, the lock is still used when the effect state's
deviceUpdate method is called to prevent asynchronous calls to reset the device
from interfering. This can be fixed by using the list lock in ALc.c instead.
Secondly, old effect states aren't immediately deleted when the effect type
changes (the actual type, not just its properties). This is because the mixer
thread is intended to be real-time safe, and so can't be freeing anything. They
are cleared away when updates reuse the container they were kept in, and they
don't incur any extra processing cost, but there may be cases where the memory
is kept around until the effect slot is deleted.
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