At 12:15 17-12-18, Politis Archontis wrote:
>Another very sensible approach was presented by Cristoff Faller and
>Illusonics in the same conference, in a simpler adaptive filter is
>used to align the microphone signals to the phase of one of the
>capsules, making them again in essence coincident.
I am interested to know how this approach gets around the problem
that with any pair of capsules, the difference in phase is a function
of the angle of incidence and how it distinguishes between identical
frequency components that are actually generated by different acoustic sources.
It seems to me that some assumptions must be involved, e.g.:
-- that the microphone array is stationary during the time of the
capture window and subsequent computation.
-- that the filtered frequency components used for the computation
contain only signals from a unique direction.
The first of these assumptions has been mentioned here by reference
to the generation of spurious signals when the mic is in motion. The
implication would be that the time taken to acquire and compute the
data is too long to satisfy the condition.
The second is demonstrably untrue when we are talking about tonal
music in a reverberant acoustic. How does the system distinguish
between a 1kHz partial from a source or reflecting surface on one
side of the array and one from a different source or reflecting
surface arriving from another direction? (A moment's thought shows
that a C major chord, distributed among various performers would have
a large number of Cs, Es, G, etc coming from all around.)
David
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