A method for auralizing a multi-microphone device. Path information for one or more sound paths using dimensions and room reflection coefficients of a simulated room for one of a plurality of microphones included in a multi-microphone device is determined. An array-related transfer functions (ARTFs) for the one of the plurality of microphones is retrieved. The auralized impulse response for the one of the plurality of microphones is generated based at least on the retrieved ARTFs and the determined path information.

Focusing sound signals in a shared 3D space uses an array of physical microphones, preferably disposed evenly across a room to provide even sound coverage throughout the room. At least one processor coupled to the physical microphones does not form beams, but instead preferably forms 1000's of virtual microphone bubbles within the room. By determining the processing gains of the sound signals sourced at each of the bubbles, the location(s) of the sound source(s) in the room can be determined. This system provides not only sound improvement by focusing on the sound source(s), but with the advantage that a desired sound source can be focused on more effectively (rather than steered to) while un-focusing undesired sound sources (like reverb and noise) instead of rejecting out of beam signals. This provides a full three dimensional location and a more natural presentation of each sound within the room.

Estimating the field strength from an ultrasonic phased array can be done by summing the contribution of each transducer to the point of interest. Since this contribution is already calculated when creating a converging spherical wave, it can be reused to add a virtual microphone to the system. By monitoring this microphone and moving it along with new focus points, a robust system of field estimates and regulation may be established.

A method, apparatus and computer program product provide an improved filter calibration procedure to reliably equalize the long term spectrum of the audio signals captured by first and second microphones that are at different locations relative to a sound source and/or are of different types. In the context of a method, the signals captured by the first and second microphones are analyzed. The method also determines one or more quality measures based on the analysis. In an instance in which one or more quality measure satisfy a predefined condition, the method determines a frequency response of the signals captured by the first and second microphones. The method also determines a difference between the frequency response of the signals captured by the first and second microphones and processes the signals captured by the first microphone for filtering relative to the signals captured by the second microphone based upon the difference.

The disclosure relates to a system, comprising a first circuit board comprising a control circuitry configured to control a beamforming array of microphones, and multiple second circuit boards attached to the first circuit board, each one of the multiple second circuit boards mounting a microphone of the array of microphones.

A transfer function calculation unit is configured to calculate a transfer function from a sound source installed in a predetermined target direction to each microphone of a microphone array, and a determination unit is configured to determine whether or not the microphone array is normal on the basis of a difference amount between a transfer function to each microphone and a predetermined ideal transfer function to each microphone.

A hearing instrument microphone device incudes at least two microphone sound ports (or sound inlets), a pressure difference microphone in communication with at least two of the sound ports and a pressure microphone in communication with at least one of the sound ports, wherein the acoustic centers of the pressure difference microphone and the pressure microphone essentially coincide.

An embodiment of the invention provides a calibration control method performed by a microphone system. The microphone system includes a plurality of microphones configured to generate a plurality of microphone signals. First, the microphone system equalize the microphone signals to generate a plurality of equalized microphone signals. Next, the microphone system calculates a set of similarity indicators based on the equalized microphone signals. Then, the microphone system compares the set of similarity indicators with a set of predetermined thresholds to determine whether to calibrate the microphone signals.

Certain aspects of the present disclosure provide an apparatus. The apparatus comprises a support structure comprising at least one microphone sensor, and a first material layer disposed adjacent to the support structure, wherein a first layer of air is formed between the first material layer and the support structure, the first layer of air being adjacent to the microphone sensor. In certain aspects, multiple material layers may be used, each of the material layers forming a layer of air. For instance, the apparatus may also include a second material layer disposed adjacent to the first material layer, wherein a second layer of air is formed between the first material layer and the second material layer.

Hearing aid apparatus includes a case, which is configured to be physically fixed to a mobile telephone. An array of microphones are spaced apart within the case and are configured to produce electrical signals in response to acoustical inputs to the microphones. An interface is fixed within the case. Processing circuitry is fixed within the case and is coupled to receive and process the electrical signals from the microphones so as to generate a combined signal for output via the interface.