There are no known file formats intended to support 3- and 4-channel UHJ, but
it is possible to store them in various audio files when a player/decoder is
aware of what it's dealing with. So there's no reason not to have it as an
option.
Currently only supports 2-channel UHJ, and the produced .amb files shouldn't be
played as normal B-Format (decoded 2-channel UHJ needs to use different shelf
filters).
This should help avoid destructive phase interference. The occlusion low-pass
filter is still applied in the time domain due to no clear topology (cutoff
frequency, slope, bandwidth, etc).
This should help improve memory use a bit since the SOFA file can be unloaded
before allocating some temp buffers for onset detection and FFT calculation.
This should produce far better results given it works directly on the frequency
response magnitudes prior to phase reconstruction, as it doesn't deal with a
linear phase filter on a truncated time-domain response (with the result also
getting truncated in both direction).
The in-library on-load HRTF resampler still uses the linear filter due to its
relative performance and simplicity benefits. It's good enough as a backup,
though users with custom HRTFs would benefit from resampling when creating the
mhr (adjusting its window size as appropriate/desired).
OpenAL-Soft fails to compile on some GCC version:
https://travis-ci.com/github/ArthurSonzogni/smk/jobs/318304162
It has regressed likely after:
c83609277b
va_start is defined in:
- <stdarg.h>
- <cstdarg>
The repository is using it from:
- al/error.cpp
- al/filter.cpp
- alc/alu.cpp
- alc/helpers.cpp
- common/alexcpt.cpp
- utils/makemhr/loaddef.cpp
- utils/openal-info.c
This patch is adding its definition in the files missing it:
- common/alexcpt.cpp
- utils/makemhr/loaddef.cpp
- utils/openal-info.c
Bug: https://travis-ci.com/github/ArthurSonzogni/smk/jobs/318304162
While maybe not technically correct, we actually only care about the difference
between onsets (any base constant is removed). This should work better since it
determines when the IR is most audible, whereas previously it used a variable
threshold of when it reached 15% of the max amplitude.
An even better method may be to check where the IR amplitude exceeds a fixed
threshold (i.e. the same threshold applied to all IRs), but that would need
tweaking to find a level that doesn't catch random noise and doesn't
potentially miss the more occluded IRs.
The generated third-order matrix has incorrect first-order coefficients,
indicating a wonky decoder. The generated second-order matrix looks more
stable.
This simply omits the scale factor from the filter, similar to how up-sampling
does. The consequence of this is less smooth transitions when ramping the
pitch while down-sampling, but otherwise behaves fine.
