e3e4dbbccf
Make the backend list static
2008-02-11 17:24:04 -08:00
7fda645b16
Call InitAL at the start of some more ALC functions
2008-02-11 17:23:31 -08:00
ec7f20644d
Fast float-to-int function is no longer needed
2008-02-08 21:03:48 -08:00
3d5fa91703
Remove unnecessary casting
2008-02-08 21:01:05 -08:00
e8acfb069c
Remove explicit dependancy on ole32 and unused dxguid
2008-02-08 20:56:45 -08:00
4c38ed272d
Enumerate DirectSound devices
2008-02-08 20:46:34 -08:00
acbb4f8e6c
Include alext.h instead of redefining some enums
2008-02-08 19:55:51 -08:00
13a3fe8a7a
Remove unneeded device struct member
2008-02-08 19:41:47 -08:00
3dd3cd4ceb
Prevent overflow of the device lists
2008-02-08 15:33:26 -08:00
80fcbc8adc
Use all capture devices listed by ALSA, not just the first on each card
2008-02-08 15:22:17 -08:00
33680accea
Don't remove the devices from the device list on unexpected shutdown
...
The close functions will remove it for us
2008-02-08 14:44:05 -08:00
081e593e2b
Set the done flag immediately when entering the initialization
...
To prevent two threads from initializing at the same time (not that it's
likely to happen at this point).
2008-02-06 22:22:29 -08:00
b4ffdfab81
Add an option for duplicating stereo sources on the back speakers
2008-02-06 22:18:50 -08:00
1ea4aac799
Add an alext.h header
2008-02-06 20:53:00 -08:00
4d6c292757
Fix race condition when starting winmm message poll thread
2008-02-03 00:28:00 -08:00
655a83c5c5
Use WAVEFORMATEXTENSIBLE for multichannel dsound output, and don't create a primary buffer
2008-02-01 08:17:57 -08:00
e8b576bc25
Use the correct channel ordering for Windows
2008-01-27 07:04:13 -08:00
01404ed7af
Fix output channel order for 6.1 and 7.1
2008-01-27 06:49:48 -08:00
ad6237f6f7
Fix availibility amount calculation
2008-01-26 21:10:55 -08:00
d06c64e1c4
aluBytesFromFormat returns bytes, not bits
2008-01-26 21:09:08 -08:00
b4ad3186dd
Update frame size after detecting the directsound output format
2008-01-25 17:05:14 -08:00
5f1c0450b3
Remove unnecessary Channels field
2008-01-25 16:38:37 -08:00
04b62f77e3
Set the output format according to the speaker setup reported by directsound
2008-01-25 05:02:00 -08:00
029dff3f40
Use both write pointers from the directsound buffer lock
2008-01-25 04:42:44 -08:00
342f639771
Release 1.2.218
2008-01-21 18:43:14 -08:00
b5a2baafe5
Remove effect slot thunk entry when deallocated forcefully
2008-01-21 15:34:38 -08:00
781e4e5be4
Remove an unneceesary pointer check and decrease indentation
2008-01-21 14:54:15 -08:00
f3dddb5e99
Remove unnecessary duplicate thunk lookups
2008-01-21 14:33:42 -08:00
9a5e892cad
Small formatting updates
2008-01-20 22:16:28 -08:00
0317362662
Remove duplicate function
2008-01-20 19:22:39 -08:00
6b403a76e8
Don't access ALSource for every sample mix
2008-01-20 19:20:24 -08:00
6bf6c6fb7b
More overflow protection
2008-01-20 13:27:54 -08:00
29618bee10
Prevent float samples from overflowing when converting to 16-bit
2008-01-20 00:43:02 -08:00
db80f682c9
Clean a couple debug messages
2008-01-19 21:41:09 -08:00
e1cdbac5a0
Close ALC first when exiting since devices might've been running when deleting stuff
2008-01-19 20:58:50 -08:00
543eb9d217
Don't use a multiple lists for extensions
2008-01-19 20:02:40 -08:00
bc963463f3
Add an option for setting the max number of sources
2008-01-19 19:38:06 -08:00
86b7b3d54b
Remove duplication of setting the max source count
2008-01-19 19:28:34 -08:00
799ba1f954
Use less ambiguous config file names
2008-01-19 18:18:14 -08:00
1a3e39e452
Remove some unnecessary duplicate math, which was making long lines
2008-01-19 00:49:05 -08:00
7dc73815ae
Remove some branches
2008-01-18 21:39:09 -08:00
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
1b9d740244
Remove duplicated source freeing code
2008-01-18 00:56:01 -08:00
497ada9c77
Buffer size fixes. Partially reverts the ALSA buffer size "fix"
2008-01-18 00:28:25 -08:00
73e386dc6f
Release effect slots when deleting sources
2008-01-17 21:17:08 -08:00
43cfc097de
Don't dereference ALContext if there's no context yet
...
Patch by Evgeny A. Marchenko
2008-01-17 12:57:22 -08:00
6735fc7911
Add missing config.h includes
2008-01-16 14:09:04 -08:00
be34dbe608
Don't include alAuxEffectSlot.h in alSource.h
2008-01-16 14:01:24 -08:00
70102e7cd3
Fix debug comments
2008-01-16 13:58:10 -08:00
8ad16145f6
Make sure sources are deleted with the context
2008-01-16 13:27:15 -08:00
Chris Robinson