| Commit message (Collapse) | Author | Age | Files | Lines |
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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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Also only access the activesource's source field once per update.
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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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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.
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Currently the only way SSE 4.1 is detected is by using __get_cpuid, i.e. with
GCC. Windows' IsProcessorFeaturePresent does not report SSE4.1 capabilities.
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Should give a bit more wiggle room for the gain stepping to get lower than the
silence threshold.
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They are still there for auxiliary sends. However, they should go away soon
enough too, and then we won't have to mess around with calculating extra
"predictive" samples in the mixer.
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This fades the dry mixing gains using a logarithmic curve, which should produce
a smoother transition than a linear one. It functions similarly to a linear
fade except that
step = (target - current) / numsteps;
...
gain += step;
becomes
step = powf(target / current, 1.0f / numsteps);
...
gain *= step;
where 'target' and 'current' are clamped to a lower bound that is greater than
0 (which makes no sense on a logarithmic scale).
Consequently, the non-HRTF direct mixers do not do not feed into the click
removal and pending click buffers, as this per-sample fading would do an
adequate job of stopping clicks and pops caused by extreme gain changes. These
buffers should be removed shortly.
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The purpose of this is to provide a safe way to be able to "swap" resources
used by the mixer from other threads without the need to block the mixer, as
well as a way to track when mixes have occurred. The idea is two-fold:
It provides a way to safely swap resources. If the mixer were to (atomically)
get a reference to an object to access it from, another thread would be able
allocate and prepare a new object then swap the reference to it with the stored
one. The other thread would then be able to wait until (count&1) is clear,
indicating the mixer is not running, before safely freeing the old object for
the mixer to use the new one.
It also provides a way to tell if the mixer has run. With this, a thread would
be able to read multiple values, which could be altered by the mixer, without
requiring a mixer lock. Comparing the before and after counts for inequality
would signify if the mixer has (started to) run, indicating the values may be
out of sync and should try getting them again. Of course, it will still need
something like a RWLock to ensure another (non-mixer) thread doesn't try to
write to the values at the same time.
Note that because of the possibility of overflow, the counter is not reliable
as an absolute count.
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