| Commit message (Collapse) | Author | Age | Files | Lines |
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It seems a simple scaling on the coefficients will allow first-order content to
work with second- and third-order coefficients, although obviously not with any
improved locality. That may be something to look into for the future, but this
is good enough for now.
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It's possible to calculate HRTF coefficients for full third-order ambisonics
now, but it's still not possible to use them here without upmixing first-order
content.
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This adds the ability to directly decode B-Format with HRTF, though only first-
order (WXYZ) for now. Second- and third-order would be easilly doable, however
we'd need to be able to up-mix first-order content (from the BFORMAT2D and
BFORMAT3D buffer formats) since it would be inappropriate to decode lower-order
content with a higher-order decoder.
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With HRTF mixing, certain things are mixed to virtual channels to be filtered
with HRTF later. This allows for using an 8-channel cube instead of a 14-
channel cube+diamond.
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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 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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I'm not sure exactly how I want to do this yet, but this is a good starting
point.
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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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Small tweaks to balance the left and right speakers, and change
unreasonably small values to 0.
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Currently missing the AL_ORIENTATION source property. Gain stepping also does
not work.
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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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https://www.gnu.org/licenses/old-licenses/lgpl-2.0.txt to fix the FSF' address Fix the FSF' address in the source
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Also avoid unnecessary clearing.
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table stuff
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