Patent ID: 7426464

Claim:
A method for reducing noise and interference for speech communication and speech recognition in an apparatus having a digital processing means for processing audio signals received in time domain from a plurality of microphones, said digital processing means comprising a first adaptive filter for enhancing a target signal in the audio signals and a second adaptive filter for reducing a non-target signal in the audio signals and an adaptive interference and noise suppression processor, said method comprising the steps: a) initializing and estimating parameters, said step comprising: a1) collecting a predetermined number of samples; a2) pre-emphasizing or whitening of the samples; a3) calculating total non-linear energy and average power of signal samples; a4) transforming the samples to two sub-bands through a Discrete Wavelet Transform; a5) estimating environment noise energy levels; a6) re-performing step a5) if total non-linear energy and average power of signal energy is below a first noise threshold and a second noise threshold respectively; a7) estimating Bark Scale noise; a8) distinguishing between abrupt change in environment noise and possible target signal; and a9) updating of the first and second noise thresholds and environment noise energy levels and Bark scale noise; b) determining direction of arrival of signal, testing for presence of target signal and processing by the first adaptive filter; c) rechecking signal from the first adaptive filter and reconfirming updated filter coefficients; d) testing for undesired signal, interference, and noise; and transforming these signals into the frequency domain; e) processing by the second adaptive filter and wrapping into Bark scale; and f) detecting and recovering unvoice signal, processing by adaptive interference and noise suppressor and high frequency recovery.