A method and system for directional processing of audio information

The system provides for the processing of audio information in a digital signal processor, where the audio information is either received from a microphone array A, B, C, D or applied to a loudspeaker array A, B, C, D. The digital signal processor applies a double delay-and-subtract beamforming algorithm to two non-collinear pairs of microphone outputs or loudspeaker inputs to provide an array with improved directionality and more uniform signal rejection away from the main beam of the beam formed signal. This can be further improved by tessellating multiple arrays into a macro-array and superimposing their responses. This processing algorithm is said to be computationally efficient and computationally inexpensive, simple to design and implement and to have improved white noise gain, alongside good low- and high-frequency performance.