There is provided a signal processing apparatus advantageous in terms of sound source separation performance. The signal processing apparatus includes a dividing unit configured to divide audio signal acquired by a plurality of audio acquisition units into components of a plurality of different frequency bands, and a processing unit configured to form, based on the audio signal, a plurality of directional beams having different directivities in accordance with a target direction and a target width. Each of the plurality of directional beams has directivities in different directions for the respective components of the frequency bands divided by the dividing unit.

A hearing system comprises a) at least one hearing device adapted for being worn on the head, or fully or partially implanted in the head, of a user, and b) a multitude of external, spatially separated, audio transmitters, each providing respective external electric sound signals comprising audio. The hearing system is configured to allow wireless communication, including audio communication, between the hearing device and the external audio transmitters, at least from the external audio transmitters to the at least one hearing device, to be established. The at least one hearing device comprises A) a multitude of microphones, each providing an electric input signal representative of sound; B) a beamformer filter providing a beamformed signal from the multitude of electric input signals; and C) an output unit configured to provide stimuli perceivable by the user as sound. The hearing system further comprises c) a selector/mixer for selecting and possibly mixing one or more of the electric input signals or the beamformed signal from the hearing device and the external electric signals from the audio transmitters and to provide a current input sound signal based thereon intended for being presented to the user, possibly in a further processed form. The selector/mixer is controlled by a source selection control signal provided by a source selection processor. The source selection processor is configured to determine the source selection control signal in dependence of a comparison of the beamformed signal and the external electric sound signals or processed versions thereof. A hearing device, and a method of operating a hearing system is further disclosed. The invention may e.g. be used in hearing aids, headsets, active ear protection devices, headphones, etc.

Method, apparatus, and computer-readable media to manage undesired sound sources in microphone pickup/focus zones preferably mitigates one or more of the undesired sound source(s) in a space having a plurality of microphones and at least one desired sound source. Preferably, at least one microphone input receives plural microphone input signals from the plurality of microphones in the space. Preferably, the least one processor is coupled to the at least one microphone input and receives the plural microphone input signals. Preferably, the at least one processor determines plural micro-zones in the space. Preferably, the at least one processor determines a threshold sound field level for each micro-zone based on received plural microphone input signals that correspond to the one or more undesired sound source(s). Preferably, the at least one processor recognizes a desired sound source when received plural microphone input signals exceed one or more threshold sound field level.

An electronic apparatus includes an array of microphones for detecting audible sounds generated by a patient and for generating audio information representing the detected audible sounds, a first beamformer having a first adaptability speed and configured to generate first audio information and first noise information from the audio information, a second beamformer having a second adaptability speed which is slower than the first adaptability speed, the second adaptive beamformer configured to generate second audio information and second noise information from the audio information, an audio classification unit for generating audio classification information based on the first audio information, a head movement detection unit for generating head movement information based on at least one of the second audio information, the first noise information, and the second noise information, and a diagnosis unit for determining a sleep apnea diagnosis based on the audio classification information and the head movement information.

Embodiments of systems and methods are described for determining which of a plurality of beamformed audio signals to select for signal processing. In some embodiments, a plurality of audio input signals are received from a microphone array comprising a plurality of microphones. A plurality of beamformed audio signals are determined based on the plurality of input audio signals, the beamformed audio signals comprising a direction. A plurality of signal features may be determined for each beamformed audio signal. Smoothed features may be determined for each beamformed audio signal based on at least a portion of the plurality of signal features. The beamformed audio signal corresponding to the maximum smoothed feature may be selected for further processing.

A hearing aid comprises a resulting beam former (Y) for providing a resulting beamformed signal Y.sub.BF based on first and second electric input signals IN.sub.1 and IN.sub.2, first and second sets of complex frequency dependent weighting parameters W.sub.11(k), W.sub.12(k) and W.sub.21(k), W.sub.22(k), and a resulting complex, frequency dependent adaptation parameter β(k)•β(k) may be determined as <C.sub.2*•C.sub.1>/<(|C2|.sup.2>+c), where * denotes the complex conjugation and • denotes the statistical expectation operator, and c is a constant, and wherein said adaptive beam former filtering unit (BFU) comprises a smoothing unit for implementing said statistical expectation operator by smoothing the complex expression C.sub.2*•C.sub.1 and the real expression |C.sub.2>.sup.2 over time. Alternatively, β(k) may be determined from the following expression

[00001]$\beta =\frac{w_{C.Math..Math.1}^H.Math.C_v.Math.w_{C.Math..Math.2}}{w_{C.Math..Math.2}^H.Math.C_v.Math.w_{C.Math..Math.2}},$

where w.sub.C1 and w.sub.C2 are the beamformer weights representing the first (C.sub.1) and the second (C.sub.2) beamformers, respectively, C.sub.v is a noise covariance matrix, and H denotes Hermitian transposition. Corresponding methods of operating a hearing aid, and a hearing aid utilizing smoothing β(k) based on adaptive covariance smoothing are disclosed.

A production work flow optimized multichannel virtual reality microphone that has its own rendering software allowing for the recording, rendering, and playing back of immersive, head-trackable positional audio for 360 video, gaming, and virtual reality applications. The multichannel microphone used to record multiple binaural sound perspectives, has eight microphones coupled to a rotatable disc frictionally mounted on the outside of a truncated spherical shell, an internal, detachable clamp for attaching the shell to a vertical pole stand, and easily accessible microphone output connections configured as four stereo microphone pairs, spaced closely to an average set of human ears. The microphone output connections are located on an internal support member. These are accessible through upper and lower lids and a door. Four small baffles simulating the pinna of the human ear reside on the disc, separating the paired microphones.

A microphone array system or microphone array unit for a conference system is provided that includes a front board, side walls and a plurality of microphone capsules arranged in or on the front board mountable on or in a ceiling of a conference room. The microphone array system or unit is adapted for generating a steerable beam within a maximum detection angle range. The microphone array system or microphone array unit includes a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located.

A partner microphone unit comprising a) a multitude microphones for picking up a sound from the environment providing corresponding electric input signals, each comprising a target signal component and a noise signal component; b) a multi-input unit noise reduction system for providing an estimate Ŝ of the target sound s comprising the person's voice and comprising a multi-input beamformer filtering unit coupled to said input units and configured to determine filter weights for providing a beamformed signal, wherein signal components from other directions than a direction of the target signal source are attenuated, whereas signal components from the direction of the target signal source are left un-attenuated; c) antenna and transceiver circuitry for establishing an audio link to another device; and wherein the multi-input beamformer filtering unit comprises an adaptive beamformer is provided.

A signal processing method and device are provided. At least two channel sound signals are acquired, and a frequency-domain audio signal corresponding to each channel sound signal is acquired; beam forming output signals of a beam group corresponding to an audio signal of each frequency point are acquired; an output direction of the beam group is acquired; and time-domain sound signals output after beam forming in the output direction are acquired.