Commit Graph

4237 Commits

Author SHA1 Message Date
Chris Robinson
0e8ca50d7a Stretch out some GUI elements for the decoder configurations 2017-03-01 20:53:52 -08:00
Chris Robinson
d1833c7b94 Increment MixCount in UpdateClockBase
This is to protect clocktime reads since the backend lock won't protect it.
2017-02-28 23:18:51 -08:00
Chris Robinson
521abf2e07 Dynamically allocate the channel delay buffers 2017-02-28 21:01:13 -08:00
Chris Robinson
51092a6315 Remove unused function declarations 2017-02-28 19:58:20 -08:00
Chris Robinson
6f2a30dea2 Remove an unneeded function 2017-02-28 19:48:44 -08:00
Chris Robinson
d3365f1b5b Start a ALC_SOFT_loopback2 extension
This extends the base ALC_SOFT_loopback extension with support for B-Format.
When ALC_FORMAT_CHANNELS_SOFT is set to ALC_BFORMAT3D_SOFT, then additional
attributes must be specified. ALC_AMBISONIC_LAYOUT_SOFT must be set to
ALC_ACN_SOFT or ALC_FUMA_SOFT for the desired channel layout,
ALC_AMBISONIC_SCALING_SOFT must be set to ALC_N3D_SOFT, ALC_SN3D_SOFT, or
ALC_FUMA_SOFT for the desired channel scaling/normalization scheme, and
ALC_AMBISONIC_ORDER_SOFT must be set to an integer value greater than 0 for the
ambisonic order (maximum allowed is implementation-dependent).

Note that the number of channels required for ALC_BFORMAT3D_SOFT is dependent
on the ambisonic order. The number of channels can be calculated by:
num_channels = (order+1) * (order+1); /* or pow(order+1, 2); */

In addition, a new alcIsAmbisonicFormatSupportedSOFT function allows apps to
determine which layout/scaling/order combinations are supported by the loopback
device. For example,
alcIsAmbisonicFormatSupported(device, ALC_ACN_SOFT, ALC_SN3D_SOFT, 2) will
check if 2nd order AmbiX (ACN layout and SN3D scaling) rendering is supported
for ALC_BFORMAT3D_SOFT output.
2017-02-28 19:01:48 -08:00
Chris Robinson
f8558ed2b7 Use a variable counter for an array size limit 2017-02-28 04:21:16 -08:00
Chris Robinson
1cd6617ff6 Don't use the mutex in the base getClockLatency implementation 2017-02-28 03:50:42 -08:00
Chris Robinson
52d1f7883b Print WARNs when a device or context error is generated 2017-02-27 20:59:52 -08:00
Chris Robinson
45d6c34015 Avoid standard malloc for buffer queue entries 2017-02-27 20:43:16 -08:00
Chris Robinson
a9610b3607 Use separate enums for the ambisonic channel order and normalization 2017-02-27 16:11:45 -08:00
Chris Robinson
5c859af24e Move the current buffer queue entry and play position to the voice
This has a couple behavioral changes. First and biggest is that querying
AL_BUFFERS_PROCESSED from a source will always return all buffers processed
when in an AL_STOPPED state. Previously all buffers would be set as processed
when first becoming stopped, but newly queued buffers would *not* be indicated
as processed. That old behavior was not compliant with the spec, which
unequivocally states "On a source in the AL_STOPPED state, all buffers are
processed."

Secondly, querying AL_BUFFER on an AL_STREAMING source will now always return
0. Previously it would return the current "active" buffer in the queue, but
there's no basis for that in the spec.
2017-02-27 15:35:15 -08:00
Chris Robinson
513c18fdc4 Ensure a non-playing or -paused source does not use a mixing voice 2017-02-25 18:10:09 -08:00
Chris Robinson
9539ccc18b Set CMP0020 for Qt 2017-02-25 16:29:27 -08:00
Chris Robinson
eceeabaf2f Improve handling of source state reads
This avoids using seq_cst for loading the source state when either inside the
mixer, or otherwise protected from inconsistencies with async updates. It also
fixes potential race conditions with getting the source offset just as a source
stops.
2017-02-24 01:47:34 -08:00
Chris Robinson
652ef2b7fd Remove an unused function 2017-02-23 20:40:16 -08:00
Chris Robinson
c2a79f0f7b Remove CalcXYZCoeffs and inline CalcAngleCoeffs 2017-02-23 16:44:59 -08:00
Chris Robinson
08948079e9 Alter how panpot/pair-wise panning works
This change allows pair-wise panning to mostly go through the normal ambisonic
panning methods, with one special-case. First, a term is added to the stereo
decoder matrix's X coefficient so that a centered sound is reduced by -3dB on
each output channel. Panning in front creates a similar gain response to the
typical
L = sqrt(1-pan)
R = sqrt(pan)
for pan = [0,1]. Panning behind the listener can reduce (up to) an additional
-10dB, creating a audible difference between front and back sounds as if
simulating head obstruction.

Secondly, as a special-case, the source positions are warped when calculating
the ambisonic coefficients so that full left panning is reached at -30 degrees
and full right at +30 degrees. This is to retain the expected 60-degree stereo
width. This warping does not apply to B-Format buffer input, although it
otherwise has the same gain responses.
2017-02-23 01:32:44 -08:00
Chris Robinson
0ce4c9b8fa Rename stereo-panning option to stereo-encoding
Also rename the 'paired' value to 'panpot', and make it the default.
2017-02-22 19:18:01 -08:00
Chris Robinson
d04cc28f33 Limit filter gains to -24dB 2017-02-22 18:07:41 -08:00
Chris Robinson
629980d15e Update ChangeLog 2017-02-22 16:31:24 -08:00
Chris Robinson
55e3b840b3 Reduce the default period count to 3 2017-02-22 15:56:09 -08:00
Chris Robinson
2e1f1449bc Don't remove a period from the OSS buffer
Since we're now waiting for space to be available before mixing, the mixing
buffer isn't adding another period.
2017-02-22 15:44:47 -08:00
Chris Robinson
e720faf2d4 Fix OpenSL latency calculation 2017-02-22 15:00:41 -08:00
Chris Robinson
5181e78c1e Reduce some code 2017-02-21 18:28:09 -08:00
Chris Robinson
2dd142fed0 Make the "sends" config option act as a limit
Instead of forcing the device to always use the specified send count, it simply
limits requests to it.
2017-02-21 17:23:54 -08:00
Chris Robinson
d3cc867bd4 Increase the default effect slot and send count
The default number of auxiliary effect slots is now 64. This can still be
raised by the config file without a hard maximum, but incurs processing cost
for each effect slot generated by the app.

The default number of source sends is now actually 2, as per the EFX docs.
However, it can be raised up to 16 via ALC_MAX_AUXILIARY_SENDS attribute
requests, rather than the previous 4.
2017-02-21 16:54:55 -08:00
Chris Robinson
864d5387dd Dynamically allocate the ALsource Send[] array 2017-02-21 16:31:59 -08:00
Chris Robinson
29994aa2de Interleave the voice and source property objects 2017-02-21 12:29:25 -08:00
Chris Robinson
cd24e42b3f Make the voices' Send[] array dynamically sized
The voices are still all allocated in one chunk to avoid memory fragmentation.
But they're accessed as an array of pointers since the size isn't static.
2017-02-21 11:17:47 -08:00
Chris Robinson
e0e6efbfea Print warnings about missing libraries and functions 2017-02-21 10:17:48 -08:00
Chris Robinson
bb4726d520 Avoid duplicating device buffer layout logic 2017-02-20 16:57:25 -08:00
Chris Robinson
f5e8a8c75e Remove an unused flag enum 2017-02-20 09:25:09 -08:00
Chris Robinson
23a34f732b Remove mention of the sinc8 resampler 2017-02-20 09:13:08 -08:00
Chris Robinson
5a2ef2590f Allow distance compensation for non-HQ rendering as well
It still requires a custom configuration to specify appropriate speaker
distances.
2017-02-20 09:08:57 -08:00
Chris Robinson
b23f81b686 Remove the separate surround51rear decoder option
Both 5.1 Side and Rear configurations use 'surround51' to look up the
appropriate decoder file. The decoder loader already handles mapping between
rear and side channels, so there's no need for separate options.
2017-02-19 22:59:55 -08:00
Chris Robinson
3761336e6c Apply distance compensation when writing to the output 2017-02-19 22:47:59 -08:00
Chris Robinson
9da152a9c8 Don't use periphonic FOA when the HOA decoder is not periphonic 2017-02-19 17:45:27 -08:00
Chris Robinson
d45dd9c668 Remove the sinc8 resampler option
Perf shows less than 1 percent CPU difference from the higher quality bsinc
resampler, but uses almost twice as much memory (a 128KB lookup table).
2017-02-19 16:45:17 -08:00
Chris Robinson
247f56249a Always lock the device backend before calling aluMixData 2017-02-18 17:32:07 -08:00
Chris Robinson
2448f88e70 Return some device latency by default
A device will never have 0 latency. OpenAL Soft itself uses a sample buffer
length of UpdateSize*NumUpdates, and during playback will have about
(NumUpdates-1) periods filled, more or less. Without a more accurate
measurement from the playback system, this is better than reporting 0.
2017-02-18 16:55:48 -08:00
Chris Robinson
d8c42918f4 Use select() to wait for audio with OSS and Solaris 2017-02-18 15:58:15 -08:00
Chris Robinson
909193a345 Reorganize ALvoice members
This places the Send[] array at the end of the struct, making it easier to
handle dynamically.
2017-02-15 17:40:26 -08:00
Chris Robinson
5a50c46c22 Make ALsourceProps' Send array dynamically sized
ALsourceProps' Send[] array is placed at the end of the struct, and given an
indeterminate size. Extra space is allocated at the end of each struct given
the number of auxiliary sends set for the device.
2017-02-14 19:59:39 -08:00
Chris Robinson
69dd570961 Fix build with non-C11 atomics 2017-02-13 21:30:20 -08:00
Chris Robinson
0d19a20901 Make the source state atomic
Since it's modified by the mixer when playback is ended, a plain struct member
isn't safe.
2017-02-13 21:18:18 -08:00
Chris Robinson
0324712540 Put BsincState in a generic union 2017-02-13 11:29:32 -08:00
Chris Robinson
841d0bb893 Porperly check for and use __builtin_assume_aligned 2017-02-13 07:36:49 -08:00
Chris Robinson
65f9b2792c Clean up the bsinc mixer a bit 2017-02-12 21:35:08 -08:00
Chris Robinson
27695e2b24 Add NEON-enhanced resamplers 2017-02-12 21:03:30 -08:00