Abstract: A harmonic echo power estimator estimates power of echo generated by harmonic loudspeaker nonlinearities in a user equipment having an echo path between a loudspeaker input and a microphone output. The estimator includes a frequency band mapper that maps each frequency band in a set of loudspeaker output signal frequency bands into a corresponding array of loudspeaker input signal frequency bands, where each frequency band in the set is mapped into several frequency bands in the corresponding array. A power estimator determines a power estimate of each input signal in each array of frequency bands. A power estimate combiner combines determined power estimates in each array of frequency bands into a corresponding estimate of loudspeaker input power generating harmonic loudspeaker nonlinearities. A power estimate transformer transforms the estimates of loudspeaker input power across the echo path into power estimates of the echo generated by the harmonic loudspeaker nonlinearities.

Abstract: A method for suppressing noise of a first signal captured via a primary microphone is provided. A primary and a reference microphone are arranged on a communication device to capture noise and intermittent speech. A determination is made whether the first signal comprises non-stationary signal components or substantially stationary noise, and whether the first signal comprises substantially far-field noise in case it was determined that it comprises non-stationary signal components. A noise power spectrum estimate of the first signal is updated with a stationary noise power spectrum estimate if the first signal is considered to comprise substantially stationary noise or a far-field noise power spectrum estimate if the first signal is considered to comprise substantially far-field noise. A frequency response is computed on the basis of the estimated noise power spectrum. Noise from the first signal is suppressed by applying the frequency response on the first signal.

Abstract: Method, device and computer program product for processing audio signals at the device, the device comprising an audio output for outputting audio signals. Audio signals are received at a plurality of microphones of the device. A characteristic of at least one of the audio signals received by the plurality of microphones is measured. A beamformer applies beamformer coefficients to the received audio signals, thereby generating a beamformer output. An echo canceller is applied to the beamformer output, thereby suppressing, from the beamformer output, an echo resulting from audio signals output from the audio output, wherein an operating parameter of the echo canceller is controlled based on the measured characteristic of the at least one of the audio signals received by the plurality of microphones.

Abstract: Method, device and computer program product for processing signals at the device. Signals are received, over a range of angles, at a plurality of sensors of the device, the received signals including an interfering signal received from an interfering source location. An interference delay pattern between receipt of signals at the sensors corresponding to receipt of a signal from the interfering source location is determined. A plurality of regularization signals having a delay pattern matching the determined interference delay pattern are generated. The generated regularization signals are used to determine beamformer coefficients to be applied by a beamformer, and the beamformer applies the determined beamformer coefficients to the signals received by the plurality of sensors, thereby generating a beamformer output.

Abstract: Beamformer coefficients may include a plurality of sets of theoretical statistical data for theoretical signals. Each theoretical signal may have its own particular attributes. The statistical data may be used in computing beamformer coefficients for application by a beamformer to signals received at a device. Signals are received at an input of the device. A respective plurality of weights is determined, for the theoretical statistical data sets, based on an analysis of the extent to which the signals have the particular attributes of the theoretical signals. The theoretical statistical data sets are retrieved, and a statistical data set is calculated for the signals by performing a weighted sum of the theoretical statistical data sets using the determined respective plurality of weights. Beamformer coefficients are computed based on the calculated statistical data set for the signals, which are used by a beamformer to the signals for generating a beamformer output.

Abstract: Methods for filtering an input signal x(k) to produce an output signal y(k) such that the ratio of a power level of the output signal to a power level of the input signal is substantially equal to a desired value ? are provided.

Abstract: One aspect of the invention provides a method for enhancing speech output by an electro-acoustical transducer in a noisy listening environment. In some embodiments, this method includes: filtering an input audio signal x(t) using a filter H(z) to produce a filtered audio signal x(t) formula (I), wherein x(t) formula (I)—H(z)x(t); providing to an electro-acoustical transducer a signal corresponding to the filtered audio signal x(t) formula (I) to produce a sound wave corresponding to the filtered audio signal; and prior to filtering the audio signal using the filter, configuring the filter such that, with respect to one or more frequencies, the filtered audio signal has a higher signal level than the input audio signal, and such that the overall signal level of the filtered audio signal (slƒ) is substantially related to the overall signal level of the input signal (slr) such that si/=sllx c.