| Commit message (Collapse) | Author | Age | Files | Lines |
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Updating and processing still differs depending on whether standard or EAX
reverb is used or not. The only functional difference should be that the
allocated buffer (and subsequent offsets) take into account the modulation
and echo times.
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They still only hold AL_TRUE or AL_FALSE, but some systems can't properly
handle atomic swaps of 1-byte types.
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Updates when the slot changes effect type is still sychronous, however, to
ensure a proper state for the Process method call. Fixing this would
essentially require all effects to work from the same state.
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The types and enums are in efx.h, which is included by alext.h. It's done this
way because EFX has a lot if definitions which would polute alext.h
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The effect state's update method will be called afterwards
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This allows the effect Update functions to handle the playback frequency being
changed. By default the effects assume a maximum frequency of 192khz, however,
it can go higher at the cost of the sample buffers being cleared and the risk
of an abort() if reallocation fails
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This should make it easier to support multiple slots
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Code created and graciously provided by Christopher Fitzgerald
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This one using the Butterworth IIR filter design
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Here is a quick description of how the reverb effect works:
+--->---+*(4)
| V new sample
+-----+---+---+ |
|extra|ltr|ref| <- +*(1)
+-----+---+---+
(3,5)*| |*(2)
+-->|
V
out sample
1) Apply master reverb gain to incoming sample and place it at the head of the
buffer. The master reverb gainhf was already applied when the source was
initially mixed.
2) Copy the delayed reflection sample to an output sample and apply the
reflection gain.
3) Apply the late reverb gain to the late reverb sample
4) Copy the end of the buffer, applying a decay gain and the decay hf ratio,
and add to the late reverb.
5) Copy the late reverb sample, adding to the output sample.
Then the head and sampling points are shifted forward, and done again for each
new sample. The extra buffer length is determined by the Reverb Density
property. A value of 0 gives a length of 0.1 seconds (long, with fairly
distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos).
The decay gain is calculated such that after a number of loops to satisfy the
Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to
the resulting output, and only getting further reduced). It is calculated as:
DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength));
Things to note: Reverb Diffusion is not currently handled, nor is Decay HF
Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this
method likely sucks, but it's the best I can come up with before release. :)
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