Commit Graph

80 Commits

Author SHA1 Message Date
Chris Robinson
ef59901e7c Set FPU mode to round toward zero for mixing 2008-08-08 07:32:21 -07:00
Chris Robinson
cfe620ccb5 Remove unnecessary casting 2008-08-08 00:21:25 -07:00
Chris Robinson
453b015225 Prevent a 0 or negative increment for the buffer position
Thanks to Christopher Fitzgerald for pointing these last two problems out
2008-08-05 20:51:30 -07:00
Chris Robinson
d3e5fcd13e Fix some calculations for the reverb buffer 2008-07-26 01:57:04 -07:00
Chris Robinson
3e0f9cc716 Make the filter processing function inline 2008-07-26 00:58:54 -07:00
Chris Robinson
c7e49c9f57 Implement yet another low-pass filter
This one using the Butterworth IIR filter design
2008-07-25 19:31:12 -07:00
Chris Robinson
559c786d0c Specify padding per buffer, and make sure it's large enough for the filter step 2008-07-24 00:41:25 -07:00
Chris Robinson
c3a7480961 Don't advertise extra samples for mixing 2008-07-23 23:27:38 -07:00
Chris Robinson
a75e75aef5 Implement an alternative low-pass filter
This method samples from the buffer so that it gets a time-correct 5khz stream,
which is subtracted from the original sample and has the high-frequency gain
applied, then added back.
A better method may be to average all the samples from the current one to the
one freq/5000 away, instead of bilinear filtering the two nearest freq/5000
apart. Processing cost will need to determine its viability
2008-07-23 22:29:53 -07:00
Chris Robinson
0042b1f80d Implement doppler factor source property 2008-07-15 02:33:05 -07:00
Chris Robinson
6d416ee734 Add the reverb room rolloff to the source room rolloff, not override 2008-07-15 02:23:53 -07:00
Chris Robinson
f369be148f Reduce the mix buffer sizes by half
Nearly 3MB is a bit much. Could reduce it further, but this is good enough for now.
2008-07-08 19:37:14 -07:00
Chris Robinson
3a09e446b3 Leave SourceToListener untransformed for use with untransformed velocities
Distance is also left untransformed so cone calculations with SoundToListener
are correct
2008-07-03 03:13:43 -07:00
Chris Robinson
fed346c285 Fix source calculations for AL_SOURCE_RELATIVE mode
Make sure the source position and direction are properly put into listener-
space before working with them, and don't calculate the listener velocity for
relative coordinates
2008-05-18 16:52:38 -07:00
Chris Robinson
49d9695ad9 Check the right struct member for the filter type 2008-04-12 07:25:18 -07:00
Chris Robinson
ec7f20644d Fast float-to-int function is no longer needed 2008-02-08 21:03:48 -08:00
Chris Robinson
3d5fa91703 Remove unnecessary casting 2008-02-08 21:01:05 -08:00
Chris Robinson
b4ffdfab81 Add an option for duplicating stereo sources on the back speakers 2008-02-06 22:18:50 -08:00
Chris Robinson
e8b576bc25 Use the correct channel ordering for Windows 2008-01-27 07:04:13 -08:00
Chris Robinson
01404ed7af Fix output channel order for 6.1 and 7.1 2008-01-27 06:49:48 -08:00
Chris Robinson
781e4e5be4 Remove an unneceesary pointer check and decrease indentation 2008-01-21 14:54:15 -08:00
Chris Robinson
f3dddb5e99 Remove unnecessary duplicate thunk lookups 2008-01-21 14:33:42 -08:00
Chris Robinson
9a5e892cad Small formatting updates 2008-01-20 22:16:28 -08:00
Chris Robinson
0317362662 Remove duplicate function 2008-01-20 19:22:39 -08:00
Chris Robinson
6b403a76e8 Don't access ALSource for every sample mix 2008-01-20 19:20:24 -08:00
Chris Robinson
1a3e39e452 Remove some unnecessary duplicate math, which was making long lines 2008-01-19 00:49:05 -08:00
Chris Robinson
7dc73815ae Remove some branches 2008-01-18 21:39:09 -08:00
Chris Robinson
4caf2c7edd Implement AL_EFFECT_REVERB
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. :)
2008-01-18 21:25:40 -08:00
Chris Robinson
43cfc097de Don't dereference ALContext if there's no context yet
Patch by Evgeny A. Marchenko
2008-01-17 12:57:22 -08:00
Chris Robinson
be34dbe608 Don't include alAuxEffectSlot.h in alSource.h 2008-01-16 14:01:24 -08:00
Chris Robinson
4742dedb45 Don't clamp wet gain if there's no send slot, and move slot gain calculation
To remove an extra if check
2008-01-16 13:00:35 -08:00
Chris Robinson
10a9bc62bf Store a reference to the effect slot in a source's send, not a copy 2008-01-16 12:43:25 -08:00
Chris Robinson
24f433b938 Remove unneeded variables 2008-01-15 21:57:50 -08:00
Chris Robinson
abc69dd3d0 Use acosf when available 2008-01-15 21:23:14 -08:00
Chris Robinson
03ca50fa70 Use the previous low-pass filter again, as it seems to match the intended output better 2008-01-15 18:29:21 -08:00
Chris Robinson
df07e8a65b Add support for AL_LOKI_quadriphonic 2008-01-14 16:11:15 -08:00
Chris Robinson
38db8eb64b Reorder setting of some variables 2008-01-12 07:36:22 -08:00
Chris Robinson
3bbbf8a025 Merge branch 'master' into efx-experiment 2008-01-11 17:19:08 -08:00
Chris Robinson
042ec206e7 Disable fast float-to-int hack.
Even with precautions, it's giving problems. Not worth it since I don't quite
understand how it works, or know if there's even a benefit.
2008-01-05 05:03:31 -08:00
Chris Robinson
312108a0d3 Try a different low-pass filter
Seems to be more correct, although it's not as powerful as the previous (which
may be a good thing)
2008-01-05 03:51:24 -08:00
Chris Robinson
5e48be27b8 Merge branch 'master' into efx-experiment 2008-01-04 14:40:38 -08:00
Chris Robinson
b3badbf97d Use 6 point spatialization for 6.1 and 7.1 output 2008-01-04 14:15:55 -08:00
Chris Robinson
8fe39042da Add the Bauer stereophonic-to-binaural DSP (bs2b) code and hooks 2008-01-03 05:36:51 -08:00
Chris Robinson
9ed574b399 Merge branch 'master' into efx-experiment 2008-01-01 06:29:11 -08:00
Chris Robinson
733cd120b3 Fix channel ordering for multichannel buffers 2008-01-01 06:16:19 -08:00
Chris Robinson
4e2f8e305e Fix wet volumes for multichannel path 2007-12-31 19:40:24 -08:00
Chris Robinson
3d78d93b40 Merge branch 'master' into efx-experiment 2007-12-31 19:34:52 -08:00
Chris Robinson
5a2f509104 Zero out wet send params when calculating source params
Instead of using a check in the mix loop
2007-12-31 19:13:18 -08:00
Chris Robinson
cb1d62f254 Add paths for 4 to 7.1 channel buffer mixing 2007-12-31 04:50:34 -08:00
Chris Robinson
1cbd625b4e Disable unnecessary calculations 2007-12-31 03:45:26 -08:00