AerialX/openal-soft-android
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215 Commits

Author SHA1 Message Date
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
1b9d740244 Remove duplicated source freeing code 2008-01-18 00:56:01 -08:00
Chris Robinson
497ada9c77 Buffer size fixes. Partially reverts the ALSA buffer size "fix" 2008-01-18 00:28:25 -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
6735fc7911 Add missing config.h includes 2008-01-16 14:09:04 -08:00
Chris Robinson
be34dbe608 Don't include alAuxEffectSlot.h in alSource.h 2008-01-16 14:01:24 -08:00
Chris Robinson
8ad16145f6 Make sure sources are deleted with the context 2008-01-16 13:27:15 -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
b95fcf5da1 Store effect slots in the context 2008-01-15 16:22:39 -08:00
Chris Robinson
df07e8a65b Add support for AL_LOKI_quadriphonic 2008-01-14 16:11:15 -08:00
Chris Robinson
729f076c3b Reduce indentation 2008-01-14 15:38:15 -08:00
Chris Robinson
dfc0118b8b Add an option for disabling ALSA mmap 2008-01-14 15:30:52 -08:00
Chris Robinson
d9ef062caf Avoid busy waiting when waiting for suspend to clear 2008-01-14 13:23:49 -08:00
Chris Robinson
2b8ce3b4cf Let the mmap thread start the pcm stream when it's full 
Instead of filling it with silence and starting it before the thread is active
 2008-01-14 13:07:41 -08:00
Chris Robinson
a27b855a39 Make sure the stream is playing when it's full 2008-01-14 12:59:44 -08:00
Chris Robinson
1634b69faf Don't restart the stream right after preparing it 
It needs to be filled, first
 2008-01-14 12:51:36 -08:00
Chris Robinson
59fa1f90d5 Don't attempt to recover the ALSA stream when filling silence 
Since it was just prepared, any error would likely mean bigger problems
 2008-01-14 12:49:21 -08:00
Chris Robinson
7a4870bd97 Close dangling devices when exiting 2008-01-14 10:54:33 -08:00
Chris Robinson
a552e32a9a Destroy context if closing a device with one 2008-01-14 10:42:11 -08:00
Chris Robinson
29c6238b52 Keep track of open devices 2008-01-14 10:39:54 -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
978764cb6b Don't limit output wave filename size 2008-01-11 15:27:56 -08:00
Chris Robinson
893ecf1af2 Fix the Wave Writer's reliance on ftell 
So output can work on FIFOs
 2008-01-11 15:18:26 -08:00
Chris Robinson
aa453b4e9e Use Sleep instead of usleep 2008-01-11 14:55:35 -08:00
Chris Robinson
f8089d2026 Don't double-close a handle on error 2008-01-11 13:23:37 -08:00
Chris Robinson
40241b4e97 Don't attempt to open ALSA playback/capture if it didn't load 2008-01-11 13:00:30 -08:00
Chris Robinson
f10408739e Add a wave file writing backend 2008-01-11 09:32:22 -08:00
Chris Robinson
bc56c00a9a Allow querying of ALC_MONO_SOURCES and ALC_STEREO_SOURCES 
Based on a patch by Xavier Bouchoux
 2008-01-10 08:24:23 -08:00
Chris Robinson
1e3ad2f9ce Use a more reliable thread loop for DSound, instead of a Win32 timer 
Also use a simpler method for calculating the read/write location
 2008-01-08 07:09:25 -08:00
Chris Robinson
9c97f07ec9 Remove unneeded silence field 2008-01-07 00:26:10 -08:00
Chris Robinson
90d825e7f8 Fix lone ALC_REFRESH query case 2008-01-06 03:36:01 -08:00
Chris Robinson
1178e900eb Don't allow 0 periods 2008-01-06 01:27:26 -08:00
Chris Robinson
8553fb9e30 ALC_REFRESH is the number of updates per second 2008-01-06 01:14:09 -08:00
Chris Robinson
da3b270488 Make OSS's update size dynamic 2008-01-06 00:19:28 -08:00
Chris Robinson
dd60366aec Fix the buffer size so ALSA doesn't multiply by the number of periods 2008-01-06 00:18:06 -08:00
Chris Robinson
017fc93307 Some non-mmap ALSA fixes 2008-01-05 05:33:54 -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
4d5885e27b Implement a crossfeed config option 2008-01-03 06:02:06 -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
7ef623c71d Fail if OSS can't set the requested bit depth and channel count 2008-01-01 06:25:00 -08:00