| Commit message (Collapse) | Author | Age | Files | Lines |
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And rename alcRing.c to ringbuffer.c for consistency.
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Also don't use inheritance with FPUCtl.
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And rename alcConfig.c to alconfig.c for consistency.
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Given that they're nearly identical, it should be relatively simple to use the
same effect state to process either of them, similar to the reverbs. The big
differences seem to be the delay range (much shorter with flanger) and the
defaults.
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Turns out the C version of the cubic resampler is just slightly faster than
even the SSE3 version of the FIR4 resampler. This is likely due to not using a
64KB random-access lookup table along with unaligned loads, both offseting the
gains from SSE.
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This will be to allow buffer layering, multiple buffers of the same format and
sample rate that are mixed together prior to resampling, filtering, and
panning. This will allow composing sounds from individual components that can
be swapped around on different invocations (e.g. layer SoundA and SoundB on one
instance and SoundA and SoundC on a different instance for a slightly different
sound, then just SoundA for a third instance, and so on). The longest buffer
within the list item determines the length of the list item.
More work needs to be done to fully support it, namely the ability to specity
multiple buffers to layer for static and streaming sources. Also the behavior
of loop points for layered static sources should be worked out. Should also
consider allowing each layer to have a sample offset.
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This reports the same ALSOFT version as alGetString(AL_VERSION), but doesn't
require a current context (which requires a ALCdevice) to call. Do *NOT* use
this version to determine feature support, use the standard interfaces. If you
think you need to use this, you probably don't, and shouldn't.
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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.
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Also keep all free property update structs together in the context instead of
per-object.
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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.
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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.
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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.
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Also rename the resampler functions to remove the unnecessary '32' token.
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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.
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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.
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