| Commit message (Collapse) | Author | Age | Files | Lines |
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This is just for the output limiter right now, but in the future can be used
for the compressor EFX effect. The parameters are also hardcoded, but can be
made configurable after 1.18.
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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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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 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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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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This change allows pair-wise panning to mostly go through the normal ambisonic
panning methods, with one special-case. First, a term is added to the stereo
decoder matrix's X coefficient so that a centered sound is reduced by -3dB on
each output channel. Panning in front creates a similar gain response to the
typical
L = sqrt(1-pan)
R = sqrt(pan)
for pan = [0,1]. Panning behind the listener can reduce (up to) an additional
-10dB, creating a audible difference between front and back sounds as if
simulating head obstruction.
Secondly, as a special-case, the source positions are warped when calculating
the ambisonic coefficients so that full left panning is reached at -30 degrees
and full right at +30 degrees. This is to retain the expected 60-degree stereo
width. This warping does not apply to B-Format buffer input, although it
otherwise has the same gain responses.
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Perf shows less than 1 percent CPU difference from the higher quality bsinc
resampler, but uses almost twice as much memory (a 128KB lookup table).
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Unsigned 32-bit offsets actually have some potential overhead on 64-bit targets
for pointer/array accesses due to rules on integer wrapping. No idea how much
impact it has in practice, but it's nice to be correct about it.
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The combined source and listener gains now can't exceed a multiplier of 16
(~24dB). This is to avoid mixes getting out of control with large volume
boosts, which reduces the effective precision given by floating-point.
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