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.
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.
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.
I don't like this, but it's currently necessary. The problem is that the
ambisonics-based panning does not maintain consistent energy output, which
causes sounds mapped directly to an output channel to be louder compared to
when being panned. The inconcistent energy output is partly by design, as it's
trying to render a full 3D sound field and at least attempts to correct for
imbalanced speaker layouts.
Although it is more correct for preserving the apparent volume, the ambisonics-
based panning does not work on the same power scale, making it louder by
comparison.
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/
At 0 distance from the listener, the sound is omni-directional. As the source
and listener become 'radius' units apart, the sound becomes more directional.
With HRTF, an omni-directional sound is handled using 0-delay, pass-through
filter coefficients, which is blended with the real delay and coefficients as
needed to become more directional.
