A vehicle microphone system may include at least two microphones forming a microphone array, at least one loudspeaker configured to emit audio signals. a processor coupled to a memory and programmed to receive incoming audio signals from the microphone array, determine at least one parameter for each channel of the microphone array, determine at least one filter to apply to at least one channel based on a difference between the parameters of each channel, and store the at least one filter in the memory.

A method is disclosed, the method comprising obtaining at least one first information indicative of audio data gathered by at least one first microphone, and at least one second information indicative of audio data gathered by at least one second microphone; determining a differential information indicative of one or more differences between at least two pieces of information, wherein the differential information is determined based, at least in part, on the at least one first information and the at least one second information; and compensating of an impact onto the audio data, wherein audio data of the first information and/or the second information is compensated based, at least in part, on the determined differential information. Further, an apparatus, and a system are disclosed.

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.

An electronic device or method for adjusting a gain on a voice operated control system can include one or more processors and a memory having computer instructions. The instructions, when executed by the one or more processors causes the one or more processors to perform the operations of receiving a first microphone signal, receiving a second microphone signal, updating a slow time weighted ratio of the filtered first and second signals, and updating a fast time weighted ratio of the filtered first and second signals. The one or more processors can further perform the operations of calculating an absolute difference between the fast time weighted ratio and the slow time weighted ratio, comparing the absolute difference with a threshold, and increasing the gain when the absolute difference is greater than the threshold. Other embodiments are disclosed.

According to certain embodiments, a microphone array having a plurality of microphone elements is calibrated by ensonifying the microphone array at a first direction relative to the microphone array with a first acoustic signal to concurrently generate a first set of audio signals from two or more of the microphone elements and processing the first set of audio signals to calibrate the two or more microphone elements. One or more other sets of audio signals can be generated by ensonifying the microphone array with one or more other acoustic signals at one or more other directions relative to the microphone array, where the two or more microphone elements are calibrated using the first set and the one or more other sets of audio signals. The calibration process can be performed outside of an anechoic chamber using one or more acoustic sources located outside or inside the microphone array.

A microphone array is configured to select an optimal pickup pattern of a microphone array including at least two microphones each associated with a predefined microphone pickup pattern. The optimal pickup pattern is selected from a list of predefined pickup patterns including the predefined microphone pickup patterns and predefined mixed pickup patterns. A method of selecting an optimal pickup pattern includes receiving microphone audio signals from the microphones to establish a plurality of microphone audio signals and mixing them in accordance with the plurality of predefined mixed pickup patterns to establish a plurality of mixed audio signals. Individual level characteristics of respective individual microphone audio signals and of respective individual mixed audio signals are determined, and an optimal pickup pattern is selected from the list of predefined pickup patterns based on the individual level characteristics.

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.

Audio device and method for operating an audio device is disclosed, the audio device comprising an interface, memory, and a processor, wherein the processor is configured to: obtain a first microphone input signal and a second microphone input signal; process the first microphone input signal and the second microphone input signal for provision of an output signal; and output the output signal. To obtain the first microphone input signal and the second microphone input signal comprises to: obtain a first microphone signal and a second microphone signal; determine a gain compensation scheme based on the first microphone signal and the second microphone signal; and compensate a gain of one or both of the first microphone signal and the second microphone signal in accordance with the gain compensation scheme for provision of the first microphone input signal and the second microphone input signal; wherein to determine the gain compensation scheme comprises to: apply a plurality of test compensation schemes to the first microphone signal and the second microphone signal; determine a performance parameter for each of the test compensation schemes; and select the gain compensation scheme based on the performance parameters.

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.

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.