Directional audio signal processing using an oversampled filterbank
a filterbank and audio signal technology, applied in the field of audio signal processing applications, can solve the problems of requiring more processing, poor b>5/b> frequency resolution, and time-domain approaches that generally provide poor performan
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[0030] Turning now to FIG. 4 an adaptive beamformer system embodying the invention in block diagram form is shown. Note that it is assumed that the outputs of the I, microphones 400 (L≧2) are already converted to digital form by a set of analogue-to-digital converters (ADC) (not shown). Similarly, the output is assumed to be converted from digital form by an digital-to-analogue converter (DAC) (not shown) to produce an appropriate output signal 490. The digitized outputs of the L microphones 400 are first combined in a combination matrix 415. The combination matrix 415 can be any Finite Impulse Response (FIR) filter with multiple input and outputs (the number of outputs M being less or equal to the number of inputs L (M≦L)). Suitable matrices include a delay-and-sum network, a sigma-delta network and a one-to-one mapping of the inputs to the outputs (for example some general matrix through which L inputs are transformed into L (i.e. M=L) outputs)). The M outputs of the combination m...
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