| Commit message (Collapse) | Author | Age | Files | Lines |
|
|
|
|
|
| |
Now FuMa and ACN channel orders are required, as are FuMa, SN3D, and N3D
normalization schemes. An integer query (alcGetIntegerv) is added for the
maximum ambisonic order.
|
|
|
|
|
| |
Also keep all free property update structs together in the context instead of
per-object.
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
The context state properties are less likely to change compared to the listener
state, and future changes may prefer more infrequent updates to the context
state.
Note that this puts the MetersPerUnit in as a context state, even though it's
handled through the listener functions. Considering the infrequency that it's
updated at (generally set just once for the context's lifetime), it makes more
sense to put it there than with the more frequently updated listener
properties. The aforementioned future changes would also prefer MetersPerUnit
to not be updated unnecessarily.
|
| |
|
| |
|
|\
| |
| | |
Fix build on Gentoo FreeBSD with freebsd-lib 9.1
|
| | |
|
| |
| |
| |
| |
| | |
Specifically, the initial reverb decay as determined by the source distance,
and the reverb decayhf limit from air absorption.
|
|/ |
|
|
|
|
|
|
|
| |
Apparently there is a bug with at least MinGW-W64 where fegetenv and fesetenv
do not properly save and restore the FPU rounding mode, resulting in the
rounding mode remaining as round-to-zero after certain function calls. I do not
know if this also affects MSVC, but better safe than sorry for now.
|
|
|
|
|
| |
This likely doesn't change anything given a working optimizer, but it cleans up
the code some.
|
|
|
|
|
| |
This makes bsincgen a native tool like bin2h, so it can run automatically when
compiling.
|
| |
|
|
|
|
|
|
|
| |
This improves the transition width, allowing more of the higher frequencies
remain audible. It would be preferrable to have an upper limit of 32 points
instead of 48, to reduce the overall table size and the CPU cost for down-
sampling.
|
| |
|
| |
|
| |
|
| |
|
| |
|
| |
|
|
|
|
|
|
|
| |
This generates the filters using the proper size and scale. The 'a' divisor
should represent the +/- sample range (and thus be a whole number), with the
number of sample points being double that. Increasing the filter size to a
multiple of 4 (for SIMD) can be done by padding in 0s afterward.
|
|
|
|
|
|
| |
Despite the claim that it was an 11th order filter, the transition width was
generated by specifying 12th order. A 12th order filter would need 14 sample
points rather than the 12 it had.
|
| |
|
|
|
|
| |
Also rename the resampler functions to remove the unnecessary '32' token.
|
|
|
|
|
|
|
| |
Rather than storing individual pointers to filter, scale delta, phase delta,
and scale phase delta entries, per phase index, the new table layout makes it
trivial to access the per-phase filter and delta entries given the base offset
and coefficient count.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
The old layout separated filters, scale deltas, phase deltas, and scale phase
deltas into separate segments that each contained a numbers of scale and phase
entries, Since processing a sample needed a filter and one of each delta entry
relating to a particular scale and phase, the memory needed would be spread
across the whole table. And since subsequent samples would use a different
phase, it would jump around the table a whole lot as well.
The new layout packs the data in a way more consistent with its use. The
filters, scale deltas, phase deltas, and scale phase deltas are interleaved,
such that for a particular scale and phase, the filter and delta entries used
are contiguous. And the phase entries for a particular scale are kept together,
so the ~500 to ~1000 samples processed per source update stay within the same
3KB to 6KB area of the 70+KB table, which is much more cache friendly.
|
| |
|
| |
|
| |
|
| |
|
| |
|
|
|
|
|
|
|
| |
Currently makehrtf only handles 24-bit output, not dual-ear, and only when
given the --experimental switch. Files produced this way will not be guaranteed
future compatibility. When the mhr format is also updated with multi-distance
measurements, the experimental switch can go away.
|
| |
|
|
|
|
|
|
|
|
|
|
|
|
| |
This improves a stereo (front-left + front-right) sound "image" by generating a
front-center channel signal. Done correctly, it helps reduce the comb effects
and phase errors associated with using only two speakers to simulate center
sounds.
Note that it shouldn't be used if the front-center channel is already included
in the positional audio mix (the dialog effect is okay). In general, it may
actually be better to exclude the front-center channel from the positional
audio mix and use this to generate front-center output.
|
| |
|
|
|
|
| |
Rather than the other way around, if a device channel isn't in the channel map.
|
|
|
|
|
|
|
| |
Not all speaker kits have a front-center speaker capable of outputing full-
range content. It's best to err on the side of caution and not include front-
center for normal positional sound by default, leaving it instead for the
dedicated dialog effect.
|
| |
|
| |
|
|
|
|
| |
Also fixes DC offset removal and increases the max IR size.
|
|
|
|
|
| |
These don't exist outside OSSv4, e.g. with OSS/Free, padsp, or aoss, so no need
to be concerned.
|
| |
|
| |
|
| |
|
|
|
|
|
| |
This effectively turns a full spread source into an ambient response,
preventing such sources from being unexpectedly quiet.
|
| |
|
| |
|
| |
|
| |
|