Audio systems, methods, and processor instructions are provided that detect voice activity of a user and provide an output voice signal. The systems, methods, and instructions receive a plurality of microphone signals and combine the plurality of microphone signals according to a first combination and a second combination. The first combination produces a primary signal having enhanced response in the direction of the user's mouth, and the second combination produces a reference signal having reduced response in the direction of the user's mouth. The primary signal and the reference signal are added and subtracted to produce a summation signal and a difference signal, respectively. The summation signal and the difference signal are compares and an output voice signal is provided based upon the comparison.

Wind noise reduction in microphone signals. A first microphone signal is obtained from a first omnidirectional microphone and, contemporaneously, a second microphone signal is obtained from a second omnidirectional microphone. The first and second microphone signals are mixed to produce an output signal. Mixing involves weighting the first and second microphone signals by respective first and second signal weights to produce respective first and second weighted microphone signals, and summing the first and second weighted microphone signals together to produce the output signal. The first and second signal weights are calculated to minimise the power of the output signal.

This application relates to methods and apparatus for digital microphones. Disclosed is a digital microphone apparatus (300) for outputting a digital output signal (DATA) at a sample rate defined by a received clock signal (CLK). The apparatus includes a band splitter (302) configured to receive a microphone signal (S.sub.MD) indicative of an output of a microphone transducer and split said microphone signal into first signal path (S.sub.P1) for frequencies in a first band and a second signal path (S.sub.P2) for frequencies in a second band, the frequencies of the second band being higher than the frequencies in the first band. A modulation block (304) is configured to operate on the second signal path to apply a selective gain modulation to signals in the second signal path.

A differential microphone array includes a number (M) of microphone sensors for converting sound to a number of electrical signals, and a processor, operably coupled to the microphone sensors, to specify a target differential order (N) for the differential microphone array, and wherein M>N+1, specify a steering matrix D comprising N+1 steering vectors, calculate a respective one of a plurality of linearly specify a steering matrix D comprising N+1 steering vectors-constrained minimum variance filters based on the steering matrix, apply the respective one of the plurality of linearly-constrained minimum variance filters to a respective one of the electrical signals to calculate a respective frequency response of the electrical signals, wherein the respective frequency response comprises a plurality of components associated with a plurality of subbands, and sum the frequency responses of the electrical signals with respect to each subband to calculate an estimated frequency spectrum of the sound.

Methods and devices for reducing intermodulation distortion are described herein. In response to receiving an audio input signal, N filtered audio signals may be generated by a filter bank, corresponding to N different frequency bands. The N filtered audio signals may be delayed by a delay time D, and a determination of whether an audio frame of the filtered audio signals includes an audio event or a non-audio event. A signal level estimation may then be determined, the signal level estimation indicating an expander, a compressor, or no compression effects being present. An amount of gain is determined and applied to the delayed audio signals, which are summed across the N frequency bands to generate a full-band audio signal. In some embodiments, the full-band audio signal may be applied to a limiter to reduce any audio clipping, and a final audio signal may be generated.

An earphone signal processing method includes: a signal picked up by a first microphone of an earphone at a position close to a mouth outside an ear canal, a signal picked up by a second microphone of the earphone at a position away from the mouth outside the ear canal and a signal picked up by a third microphone are acquired, the third microphone being in a cavity formed by the earphone and the ear canal; dual-microphone noise reduction is performed on the signals picked up by the first and second microphones to obtain a first intermediate signal; dual-microphone noise reduction is performed on the signals picked up by the second and third microphones to obtain a second intermediate signal; the first and second intermediate signals are fused to obtain a fused voice signal; and the fused voice signal is output.

An image capture device includes a housing having a lens snout protruding from a front housing surface. A front microphone is mounted below the lens snout. A top microphone is mounted under a top housing surface. The top microphone is positioned to receive direct freestream air flow at a first pitched forward angle. The front microphone is positioned to receive turbulent air flow at a second pitched forward angle. The second pitched forward angle is greater than or equal to the first pitched forward angle. An audio processor receives a first audio signal and a second audio signal from the top microphone and front microphone, respectively. The audio processor generates frequency sub-bands from the first and second audio signals. The audio processor selects the frequency sub-bands with the lowest noise metric and combines them to generate an output audio signal.

In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

Disclosed herein, among other things, are methods and apparatus for a level-dependent compression system for hearing assistance devices, such as hearing aids. The present subject matter includes a hearing assistance device having a buffer for receiving time domain input signals and a frequency analysis module to convert time domain input signals into a plurality of subband signals. A power detector is adapted to receive the subband signals and to provide a subband version of the input signals. A nonlinear gain stage applies gain to the plurality of subband versions of the input signals, and a frequency synthesis module processes subband signals from the nonlinear gain stage and to create a processed output signal. The device also includes a filter for filtering the signals, and a level-dependent compression module. The level-dependent compression module is adapted to provide bandwidth control to the plurality of subband signals produced by the frequency analysis stage.