1656 Commits

Author SHA1 Message Date
Chris Robinson
14bc7baeb7 Store the source prop updates with the mixer voice
Also move its declaration and rename it for consistency.
2017-04-17 21:16:01 -07:00
Chris Robinson
660971d0b7 Close some gaps in enum values 2017-04-16 23:22:30 -07:00
Chris Robinson
064176d03d Remove some unnecessary parenthesis 2017-04-16 15:05:57 -07:00
Chris Robinson
d9bf4f7620 Allow increasing the maximum source limit
If the requested number of mono and stereo sources exceeds 256, the source
limit will be expanded. Any config file setting overrides this. If the device
is reset to have fewer sources than are currently allocated, excess sources
will remain and be usable as normal, but no more can be generated until enough
are delated to go back below the limit.
2017-04-14 23:50:49 -07:00
Chris Robinson
f94fa5d5cf Use separate atomic macros for pointers 2017-04-14 17:47:55 -07:00
Chris Robinson
9e60eea93b Use atomic flags for the thunk array 2017-04-14 16:14:05 -07:00
Chris Robinson
24c172bb96 Use ALsizei for the fir4 resampler fraction 2017-04-12 22:45:54 -07:00
Chris Robinson
901804d724 Store the ambisonic order separate from the channel enum 2017-04-12 18:26:07 -07:00
Chris Robinson
6cc69c8d94 Add a sample converter
This is intended to do conversions for interleaved samples, and supports
changing from one DevFmtType to another as well as resampling. It does not
handle remixing channels.

The mixer is more optimized to use the resampling functions directly. However,
this should prove useful for recording with certain backends that won't do the
conversion themselves.
2017-04-10 09:26:06 -07:00
Chris Robinson
81527cdbdd Convert the CoreAudio backend to the updated backend API 2017-04-09 11:21:02 -07:00
Chris Robinson
aef774a7a0 Handle the source offset fraction as an ALsizei 2017-04-08 14:29:08 -07:00
Chris Robinson
319d097198 Pre-compute the sinc4 resampler coefficient table 2017-04-08 13:43:19 -07:00
Chris Robinson
338d61f907 Reference count HRTFs and unload them when unused 2017-04-06 13:00:29 -07:00
Chris Robinson
2eaa10fc21 Load HRTF files as needed
Currently only applies to external files, rather than embedded datasets. Also,
HRTFs aren't unloaded after being loaded, until library shutdown.
2017-04-05 12:27:30 -07:00
Chris Robinson
f76dea0c03 Store the loaded hrtf entry container in the enumerated hrtf entry 2017-04-05 11:29:58 -07:00
Chris Robinson
b78ddc7ef7 Make sure the mix is done after setting the looping property 2017-04-02 06:40:56 -07:00
Chris Robinson
af833c8554 Remove a couple more uses of BYTE3 2017-03-31 09:21:31 -07:00
Chris Robinson
90c005bbec Convert float samples to integer using a power-of-2 multiple 2017-03-31 09:11:28 -07:00
Chris Robinson
355a8898cf Remove the (u)byte3 sample formats
They're not accessible since the removal of the buffer_samples extension, and
were kind of clunky to work with as 24-bit packed values.
2017-03-31 08:15:20 -07:00
Chris Robinson
ac8b4aa5f6 Convert integer samples to float using a power-of-2 divisor
This should cut down on unnecessary quantization noise (however minor) for 8-
and 16-bit samples. Unfortunately a power-of-2 multiple can't be used as easily
for converting float samples to integer, due to integer types having a non-
power-of-2 maximum amplitude (it'd require more per-sample clamping).
2017-03-31 06:54:46 -07:00
Chris Robinson
70aefa75e2 Use an array of pointers for effects instead of a linked list 2017-03-27 23:16:23 -07:00
Chris Robinson
1aca344688 Fix handling of the PropsClean flags 2017-03-23 19:32:53 -07:00
Chris Robinson
1c49d0542d Use an atomic flag to mark auxiliary effect slot updates 2017-03-23 19:16:32 -07:00
Chris Robinson
5404b2225a Add some comments for ALsource functions 2017-03-23 01:16:13 -07:00
Chris Robinson
cdfe0d8f5a Use an atomic flag to test if a source needs to update 2017-03-20 21:25:39 -07:00
Chris Robinson
e06cf07ab0 Break up a function and move the code to where it's called 2017-03-19 16:49:23 -07:00
Chris Robinson
ecfa1dcb6f Don't defer source state or offset changes 2017-03-19 13:48:40 -07:00
Chris Robinson
cdf63b553f Avoid doing sequential load for the source state 2017-03-12 19:27:51 -07:00
Chris Robinson
7b4645f5f8 Store the HRIR coeff pointer and delays directly in MixHrtfParams 2017-03-12 06:58:27 -07:00
Chris Robinson
96aaab9366 Rework HRTF coefficient fading
This improves fading between HRIRs as sources pan around. In particular, it
improves the issue with individual coefficients having various rounding errors
in the stepping values, as well as issues with interpolating delay values.

It does this by doing two mixing passes for each source. First using the last
coefficients that fade to silence, and then again using the new coefficients
that fade from silence. When added together, it creates a linear fade from one
to the other. Additionally, the gain is applied separately so the individual
coefficients don't step with rounding errors. Although this does increase CPU
cost since it's doing two mixes per source, each mix is a bit cheaper now since
the stepping is simplified to a single gain value, and the overall quality is
improved.
2017-03-11 18:04:06 -08:00
Chris Robinson
feffe1e81a Make the voice's 'moving' state a bitflag 2017-03-11 06:26:05 -08:00
Chris Robinson
98e8f941b7 Allocate as many channels for DirectHrtfState as needed 2017-03-11 06:20:04 -08:00
Chris Robinson
6b4b00e462 Dynamically allocate the device's HRTF state 2017-03-10 10:47:43 -08:00
Chris Robinson
583d431947 Implement NFC filters for Ambisonic rendering
NFC filters currently only work when rendering to ambisonic buffers, which
includes HQ rendering and ambisonic output. There are two new config options:
'decoder/nfc' (default on) enables or disables use of NFC filters globally, and
'decoder/nfc-ref-delay' (default 0) specifies the reference delay parameter for
NFC-HOA rendering with ambisonic output (a value of 0 disables NFC).

Currently, NFC filters rely on having an appropriate value set for
AL_METERS_PER_UNIT to get the correct scaling. HQ rendering uses the averaged
speaker distances as a control/reference, and currently doesn't correct for
individual speaker distances (if the speakers are all equidistant, this is
fine, otherwise per-speaker correction should be done as well).
2017-03-10 04:35:32 -08:00
Chris Robinson
9454d3e776 Move ALvoice declaration to alu.h 2017-03-09 07:23:12 -08:00
Chris Robinson
5ffb0842ac Remove unnecessary atomic members 2017-03-08 04:59:22 -08:00
Chris Robinson
b1b3a369ef Remove an unnecessary variable 2017-03-07 09:38:33 -08:00
Chris Robinson
b64da108a9 Check that a source is actually playing before setting paused
Also slightly refactor setting playing state when the device is disconnected or
there's no buffers to play.
2017-03-07 04:50:09 -08:00
Chris Robinson
190120dfd7 Store the channel count and sample size in the voice 2017-03-07 00:19:40 -08:00
Chris Robinson
bc0096365e Don't modify the source state in the mixer 2017-03-07 00:19:40 -08:00
Chris Robinson
1b3100ab9a Remove an unused function 2017-03-07 00:19:40 -08:00
Chris Robinson
073829f26a Make the voice's source pointer atomic 2017-03-05 04:50:27 -08:00
Chris Robinson
6d7a790183 Add a boolean to specify if a voice should be playing 2017-03-02 01:02:40 -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
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
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