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).
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.
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.
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.
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.
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.
This should improve positional quality for relatively low cost. Full HRTF
rendering still only uses first-order since the only use of the dry buffer
there is for first-order content (B-Format buffers, effects).
Designed for apps that either don't change the listener's AL_GAIN, or don't
allow the listener's AL_GAIN to go above 1. This allows the volume to still be
increased further than such apps may allow, if users find it too quiet.
Be aware that increasing this can easily cause clipping. The gain limit
reported by AL_GAIN_LIMIT_SOFT is also affected by this.
This appears to be how Creative's Windows drivers handle it, and is necessary
for at least the Windows version of UT2k4 (otherwise it tries to play a source
while suspended, checks and sees it's stopped, then kills it before it's given
a chance to start playing).
Consequently, the internal properties it gets mixed with are determined by what
the source properties are at the time of the play call, and the listener
properties at the time of the suspend call.
This does not change alDeferUpdatesSOFT, which will still hold the play state
change until alProcessUpdatesSOFT.
Note that this now also causes all playing sources to update when an effect
slot is updated. This is a bit wasteful, as it should only need to re-update
sources that are using the effect slot (and only when a relevant property is
changed), but it's good enough. Especially with deferring since all playing
sources are going to get updated on the process call anyway.
