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This method is intended to help development by easily testing the quality of
the B-Format encode and B-Format-to-HRTF decode. When used with HRTF, all
sources are renderer using the virtual B-Format output, rather than just
B-Format sources.
Despite the CPU cost savings (only four channels need to be filtered with HRTF,
while sources all render normally), the spatial acuity offered by the B-Format
output is pretty poor since it's only first-order ambisonics, so "full" HRTF
rendering is definitely preferred.
It's /possible/ for some systems to be edge cases that prefer the CPU cost
savings provided by basic over the sharper localization provided by full, and
you do still get 3D positional cues, but this is unlikely to be an actual use-
case in practice.
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Largely copied from JACK, it's extended to work with user-specified element
sizes instead of bytes. This is necessary to be able to work with 6- and 7-
channel output modes.
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The sound localization with virtual channel mixing was just too poor, so while
it's more costly to do per-source HRTF mixing, it's unavoidable if you want
good localization.
This is only partially reverted because having the virtual channel is still
beneficial, particularly with B-Format rendering and effect mixing which
otherwise skip HRTF processing. As before, the number of virtual channels can
potentially be customized, specifying more or less channels depending on the
system's needs.
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This way takes into account a new stereo-mode config option, which when set to
"headphones" will default to using HRTF. Eventually the device will also be
able to specify if headphones are being used.
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This new method mixes sources normally into a 14-channel buffer with the
channels placed all around the listener. HRTF is then applied to the channels
given their positions and written to a 2-channel buffer, which gets written out
to the device.
This method has the benefit that HRTF processing becomes more scalable. The
costly HRTF filters are applied to the 14-channel buffer after the mix is done,
turning it into a post-process with a fixed overhead. Mixing sources is done
with normal non-HRTF methods, so increasing the number of playing sources only
incurs normal mixing costs.
Another benefit is that it improves B-Format playback since the soundfield gets
mixed into speakers covering all three dimensions, which then get filtered
based on their locations.
The main downside to this is that the spatial resolution of the HRTF dataset
does not play a big role anymore. However, the hope is that with ambisonics-
based panning, the perceptual position of panned sounds will still be good. It
is also an option to increase the number of virtual channels for systems that
can handle it, or maybe even decrease it for weaker systems.
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Apparently, 5.1 surround sound is supposed to use the "side" channels, not the
back channels, and we've been wrong this whole time. That means the "5.1 Side"
is actually the correct 5.1 setup, and using the back channels is anomalous.
Additionally, this means the 5.1 buffer format should also use the the side
channels instead of the back channels.
A final note: the 5.1 mixing coefficients are changed so both use the original
5.1 surround sound set (with the surround channels at +/-110 degrees). So the
only difference now between 5.1 "side" and 5.1 "back" is the channel labels.
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Currently missing the AL_ORIENTATION source property. Gain stepping also does
not work.
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This behavior better matches Creative's hardware drivers and Rapture3D's OpenAL
driver. A compatibility environment variable is provided to restore the old
no-op behavior for any app that behaves badly from this change (set
__ALSOFT_SUSPEND_CONTEXT to "ignore").
If too many apps have a problem with this, the default behavior may need to be
changed to ignore, with the env var providing an option to defer/batch instead.
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For mono sources, third-order ambisonics is utilized to generate panning gains.
The general idea is that a panned mono sound can be encoded into b-format
ambisonics as:
w[i] = sample[i] * 0.7071;
x[i] = sample[i] * dir[0];
y[i] = sample[i] * dir[1];
...
and subsequently rendered using:
output[chan][i] = w[i] * w_coeffs[chan] +
x[i] * x_coeffs[chan] +
y[i] * y_coeffs[chan] +
...;
By reordering the math, channel gains can be generated by doing:
gain[chan] = 0.7071 * w_coeffs[chan] +
dir[0] * x_coeffs[chan] +
dir[1] * y_coeffs[chan] +
...;
which then get applied as normal:
output[chan][i] = sample[i] * gain[chan];
One of the reasons to use ambisonics for panning is that it provides arguably
better reproduction for sounds emanating from between two speakers. As well,
this makes it easier to pan in all 3 dimensions, with for instance a "3D7.1" or
8-channel cube speaker configuration by simply providing the necessary
coefficients (this will need some work since some methods still use angle-based
panpot, particularly multi-channel sources).
Unfortunately, the math to reliably generate the coefficients for a given
speaker configuration is too costly to do at run-time. They have to be pre-
generated based on a pre-specified speaker arangement, which means the config
options for tweaking speaker angles are no longer supportable. Eventually I
hope to provide config options for custom coefficients, which can either be
generated and written in manually, or via alsoft-config from user-specified
speaker positions.
The current default set of coefficients were generated using the MATLAB scripts
(compatible with GNU Octave) from the excellent Ambisonic Decoder Toolbox, at
https://bitbucket.org/ambidecodertoolbox/adt/
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