Abstract: Disclosed herein are systems and techniques for microphone array calibration, as well as communication systems in which calibrated microphones can be used. The systems and techniques disclosed herein may provide both phase and magnitude calibration for microphone arrays, resulting in improved performance for beamforming and other applications. Further, various systems and methods are disclosed herein for local storage of calibration coefficients in the microphone array (e.g., at the time of manufacture and calibration). Further, various systems and methods disclosed herein may include central application of the calibration of a microphone array (e.g., in an edge processor at operation time) to replace uncalibrated microphone signals with calibrated microphone signals further down in the signal chain.

Abstract: In an example, a machine-implemented method for detecting voice activity may include receiving a digital representation of an audio signal. The method may also include applying a first stage which may include determining a first frequency-domain indicator from the digital representation of the audio signal to identify a candidate speech duration. The method may also include applying a second stage which may include determining at least one of a mel-frequency cepstral (MFC) indicator or a pitch indicator from the digital representation of the audio signal to assess whether the identified candidate speech duration contains speech.

Abstract: A method and apparatus are provided for generating a directional output signal from sound received by at least two microphones arranged as microphone array. The method includes transforming the sound received by each of said microphones and represented by analog-to-digital converted time-domain signals provided by each of said microphones into corresponding complex-valued frequency-domain microphone signals each having a frequency component value for each of a plurality of frequency components, calculating, for each of the plurality of frequency components, real-valued Wind Reduction Factors as minima of the reciprocal and non-reciprocal frequency components of a plurality of real-valued Deviation Spectra if said minimum is below a preselected deviation threshold and set to one if said minimum is above or equal to said deviation threshold, and for each of the plurality of frequency components, forming a frequency-domain wind-reduced output signal.

Abstract: A method and apparatus are provided for generating a directional output signal from sound received by at least two microphones arranged as microphone array. The method includes transforming the sound received by each of said microphones and represented by analog-to-digital converted time-domain signals provided by each of said microphones into corresponding complex-valued frequency-domain microphone signals each having a frequency component value for each of a plurality of frequency components, and calculating, for each of the plurality of frequency components of the complex-valued frequency-domain microphone signal of at least one of said microphones, a respective tolerance compensated frequency component value by multiplying the frequency component value of the complex-valued frequency-domain microphone signal of said microphone with a frequency-specific real-valued correction factor.

Abstract: A method and apparatus are provided for generating a directional output signal from sound received by at least two microphones arranged as microphone array.

Abstract: A method and apparatus are provided for adaptively generating a directional output signal from sound received by at least two microphones arranged as microphone array.

Abstract: A method and apparatus are provided for generating a directional output signal from sound received by at least two microphones arranged as microphone array. The directional output signal has one or more Beam Focus Directions. The method includes transforming sound received by each microphone into a corresponding complex valued frequency-domain microphone. For any Beam Focus Direction a Beam Focus Spectrum is calculated, consisting, for each of the plurality of frequency components, of time-dependent, real-valued attenuation factors being calculated based on the plurality of microphone signals. For each of the plurality of frequency components, the maximum amongst those attenuation factors of different Beam Focus Spectra is selected and multiplied with the frequency component of the complex-valued frequency-domain signal of one microphone, forming a frequency-domain multi-focus directional output signal, from which by means of inverse transformation a time-domain signal can be synthesized.

Abstract: An audio signal processor generating an analog audio signal with directional characteristic from at least two 1-bit audio signals of digital microphones comprises a shift register of length n for bit-clocked throughput of at least one 1-bit audio signal and a combiner for combining at least two 1-bit audio signals to an interlaced 1-bit audio signal with doubled bit frequency.

Abstract: An audio signal processor generating an analog audio signal with directional characteristic from at least two 1-bit audio signals of digital microphones comprises a shift register of length n for bit-clocked throughput of at least one 1-bit audio signal and a combiner for combining at least two 1-bit audio signals to an interlaced 1-bit audio signal with doubled bit frequency.

Abstract: A method and apparatus are provided for generating a noise reduced output signal from sound received by a first microphone. The method includes transforming the sound received by the first microphone into a first input signal and transforming sound received by a second microphone into a second input signal. The method includes calculating, for each of a plurality of frequency components, an energy transfer function value as a real-valued quotient by dividing a temporally averaged product of an amplitude of the first input signal and the second input signal by a temporally averaged absolute square of the second input signal, calculating a gain value as a function of the calculated energy transfer function value, and generating the noise reduced output signal based on the product of the first input signal and the calculated gain value at each of the plurality of frequency components.

Abstract: A method and apparatus for generating a noise reduced output signal from sound received by a first and second microphone arranged as a microphone array. The method includes transforming sound received by the first microphone into a first input signal and sound received by a second microphone into a second input signal and calculating, for each of the frequency components, a weighted sum of at least two intermediate signals calculated from the input signals by means of complex valued transfer functions and real valued Equalizer functions. The method includes a weighing function with range between zero and one, with quotients of signal energies of the intermediate functions as arguments of the weighing function, and generating the noise reduced output signal based on the weighted sum of the intermediate functions and based on the weighted sum of the first and second intermediate function at each of the frequency components.

Abstract: A method and apparatus for generating a noise reduced output signal from sound received by a first and second microphone arranged as a microphone array. The method includes transforming sound received by the first microphone into a first input signal and sound received by a second microphone into a second input signal and calculating, for each of the frequency components, a weighted sum of at least two intermediate signals calculated from the input signals by means of complex valued transfer functions and real valued Equalizer functions. The method includes a weighing function with range between zero and one, with quotients of signal energies of the intermediate functions as arguments of the weighing function, and generating the noise reduced output signal based on the weighted sum of the intermediate functions and based on the weighted sum of the first and second intermediate function at each of the frequency components.

Abstract: A method and device for phase-sensitive processing of sound signals of at least one sound source may include arranging two microphones at a distance d from each other, capturing sound signals with both microphones, generating associated microphone signals, and processing the sound signals of the microphones. During a calibration mode, a calibration-position-specific, frequency-dependent phase difference vector ?0(f) between the associated calibration microphone signals may be calculated from their frequency spectra for the calibration position. Then, during an operating mode, a signal spectrum S of a signal to be output is calculated by multiplication of at least one of the two frequency spectra of the current microphone signals with a spectral filter function F.

Abstract: A method and apparatus are provided for generating a noise reduced output signal from sound received by a first microphone. The method includes transforming the sound received by the first microphone into a first input signal and transforming sound received by a second microphone into a second input signal. The method includes calculating, for each of a plurality of frequency components, an energy transfer function value as a real-valued quotient by dividing a temporally averaged product of an amplitude of the first input signal and the second input signal by a temporally averaged absolute square of the second input signal, calculating a gain value as a function of the calculated energy transfer function value, and generating the noise reduced output signal based on the product of the first input signal and the calculated gain value at each of the plurality of frequency components.

Abstract: A method and device for phase-sensitive processing of sound signals of at least one sound source may include arranging two microphones at a distance d from each other, capturing sound signals with both microphones, generating associated microphone signals, and processing the sound signals of the microphones. During a calibration mode, a calibration-position-specific, frequency-dependent phase difference vector ?0(f) between the associated calibration microphone signals may be calculated from their frequency spectra for the calibration position. Then, during an operating mode, a signal spectrum S of a signal to be output is calculated by multiplication of at least one of the two frequency spectra of the current microphone signals with a spectral filter function F.

Abstract: The invention relates to a hearing aid to be arranged at and/or in an ear, comprising a microphone for converting acoustic signals into electrical signals, a hearing module for providing the electrical signals, a loudspeaker for converting the electrical signals outputted from the hearing module into acoustic signals. The hearing module comprises a means for noise suppression effecting a noise estimation to determine a signal-dependent filter and to provide a noise-reduced output signal.

Abstract: A method and device for phase-sensitive processing of sound signals of at least one sound source may include arranging two microphones at a distance d from each other, capturing sound signals with both microphones, generating associated microphone signals, and processing the sound signals of the microphones. During a calibration mode, a calibration-position-specific, frequency-dependent phase difference vector ?0(f) between the associated calibration microphone signals may be calculated from their frequency spectra for the calibration position. Then, during an operating mode, a signal spectrum S of a signal to be output is calculated by multiplication of at least one of the two frequency spectra of the current microphone signals with a spectral filter function F.

Abstract: A method and device for phase-sensitive processing of sound signals of at least one sound source may include arranging two microphones at a distance d from each other, capturing sound signals with both microphones, generating associated microphone signals, and processing the sound signals of the microphones. During a calibration mode, a calibration-position-specific, frequency-dependent phase difference vector ?0(f) between the associated calibration microphone signals may be calculated from their frequency spectra for the calibration position. Then, during an operating mode, a signal spectrum S of a signal to be output is calculated by multiplication of at least one of the two frequency spectra of the current microphone signals with a spectral filter function F.

Abstract: Method and device for analysing and adjusting acoustic properties of a hands-free device of a motor vehicle, including at least one hands-free microphone, at least one vehicle loudspeaker and a first radio interface, using a calibrated measuring microphone and a calibrated test loudspeaker and a data processing device which is connected to the measuring microphone and test loudspeaker and includes a second radio interface.

Abstract: In a device (10, 10?) for determining the state of anchoring of an implanted endoprosthesis (12, 12?), comprising means for oscillation of the endoprosthesis (12, 12?) and means for detecting the state of oscillation of the endoprosthesis (12, 12?), it is proposed that the means for oscillation of the endoprosthesis (12, 12?) are designed to emit a modulated ultrasound signal, comprising an ultrasound carrier signal and a tunable modulation signal.