| Commit message (Collapse) | Author | Age | Files | Lines |
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This properly accounts for the room rolloff factor for normal air absorption
(which makes it none by default, like distance attenuation), and uses the
reverb's decay time, decay hf ratio, decay hf limit, and room air absorption
properties to calculate an initial hf decay with the WetGainAuto flag. This
mirrors the behavior of the initial distance decay.
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The previous value couldn't actually be expressed as a float and got rounded up
to the next whole number value, leaving the potential for an overrun in the
squared sum.
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This helps keep the squared sum stable over larger updates, also avoiding the
need to keep recalculating it.
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Though it didn't strictly clash since it was for a different component (global
state vs source property), 0x1213 was used by AL_RESAMPLER_NAME_SOFT. Probably
best to avoid duplicate property values regardless.
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This is a bit more efficient than calling the normal HRTF mixing function
twice, and helps solve the problem of the values generated from convolution not
being consistent with the new HRIR.
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This reduces the output volume when the mixed samples extend outside of -1,+1,
to prevent excessive clipping. It can reduce the volume by -80dB in 50ms, and
increase it by +80dB in 1s (it will not go below -80dB or above 0dB).
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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.
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This removes the need to access a couple more source fields in the mixer, and
also makes the looping and queue fields non-atomic.
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Also move its declaration and rename it for consistency.
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This is intended to do conversions for interleaved samples, and supports
changing from one DevFmtType to another as well as resampling. It does not
handle remixing channels.
The mixer is more optimized to use the resampling functions directly. However,
this should prove useful for recording with certain backends that won't do the
conversion themselves.
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Currently only applies to external files, rather than embedded datasets. Also,
HRTFs aren't unloaded after being loaded, until library shutdown.
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They're not accessible since the removal of the buffer_samples extension, and
were kind of clunky to work with as 24-bit packed values.
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This improves fading between HRIRs as sources pan around. In particular, it
improves the issue with individual coefficients having various rounding errors
in the stepping values, as well as issues with interpolating delay values.
It does this by doing two mixing passes for each source. First using the last
coefficients that fade to silence, and then again using the new coefficients
that fade from silence. When added together, it creates a linear fade from one
to the other. Additionally, the gain is applied separately so the individual
coefficients don't step with rounding errors. Although this does increase CPU
cost since it's doing two mixes per source, each mix is a bit cheaper now since
the stepping is simplified to a single gain value, and the overall quality is
improved.
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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).
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