Disclosed is a wearable device for producing noise free communication. The wearable device includes a housing configured to wear by a user, an air conduction microphone configured in the housing to receive voice sound of the user, an accelerometer sensor configured in the housing to receive voice signature, a battery configured in the housing to power the air conduction microphone and the accelerometer sensor, a printed circuit board configured in the housing to receive power from the battery, a memory unit connected to the printed circuitry board to store plurality of instructions, and a digital signal processor connected to the printed circuit board to process the stored plurality of instructions. The instructions are programmed to achieve a noise free communication. The voice from both air conduction microphone and the accelerometer sensor are analyzed to filter out noise and resulting in generation of noise free communication.

Systems and methods include a first voice activity detector operable to detect speech in a frame of a multichannel audio input signal and output a speech determination, a constrained minimum variance adaptive filter operable to receive the multichannel audio input signal and the speech determination and minimize a signal variance at the output of the filter, thereby producing an equalized target speech signal, a mask estimator operable to receive the equalized target speech signal and the speech determination and generate a spectral-temporal mask to discriminate a target speech from noise and interference speech, and a second activity voice detector operable to detect voice in a frame of the speech discriminated signal. An audio input sensor array including a plurality of microphones, each microphone generating a channel of the multichannel audio input signal. A sub-band analysis module operable to decompose each of the channels into a plurality of frequency sub-bands.

An audio signal processing circuit is disclosed. The audio signal processing circuit includes a digital signal processing part formed in a digital area, and configured to process a digital audio signal; an analog circuit formed in an analog area, and configured to process an analog audio signal; a frequency divider formed in the digital area, and configured to divide a system clock signal to generate a first clock signal to be provided to the digital signal processing part and a second clock signal to be provided to the analog area; and a retiming circuit formed in the analog area, and configured to retime the second clock signal by using the system clock signal and provide the retimed second clock signal to the analog circuit.

The present disclosure relates to microphones and electronic devices having the same. A microphone may include a housing for receiving vibration signals; a converting component inside the housing for converting the vibration signals into electrical signals, and a processing circuit for processing the electrical signals. The converting component may include a transducer and at least one damping film attached to the transducer.

A hearing aid includes a feedback control system for handling external feedback from an output transducer to an input transducer. The feedback control system includes an open loop gain estimator for providing an instant open loop gain estimate; an adaptive filter configured to provide a current estimate of the feedback path transfer function; a feedback change estimator configured to provide an instant estimate of the feedback path transfer function in dependence of the forward path transfer function, the instant open loop gain estimate; and an adaptive filter controller for providing an update transfer function estimate for the adaptive filter in dependence of the instant estimate of the feedback path transfer function. The hearing aid is configured to use the update transfer function estimate in the adaptive filter to update the current estimate of the feedback path transfer function. A method of detecting a sudden change in a feedback/echo path is further disclosed.

Embodiments relate generally to devices and methods for improved active noise cancellation hearing protection. For example, multiple speakers might be used in each earcup, with each speaker providing active noise cancellation for a narrow frequency range which is a portion of the total frequency range of the entire array of speakers. By dividing the active noise cancellation up across an array of speakers, each speaker can better target its specific narrow frequency range, resulting in improved active noise cancellation.

An apparatus includes a headphone driver and a processor in communication with the headphone driver. The processor is configured to receive an audio setting selection from among a plurality of audio setting selections. Each audio setting selection is associated with a frequency range that includes at least one frequency that is outside of a range of human hearing. The processor is further configured to receive an audio signal and to process the audio signal according to the selected audio setting selection to generate an output signal. The processor is configured to provide an output signal to the headphone driver.

A noise suppressing apparatus calculates a phase difference on the basis of a first and second sound signal obtained by a microphone array; calculates a first sound arrival rate on the basis of a first phase difference area and the phase difference and a second sound arrival rate on the basis of a second phase difference area and the phase difference; calculates a dissimilarity that represents a level of difference between the first sound arrival rate and the second sound arrival rate; determines whether the pickup target sound is included in the first sound signal on the basis of the dissimilarity; and determines a suppression coefficient to be applied to the frequency spectrum of the first sound signal, on the basis of a result of the determination of whether the pickup target sound is included and on the basis of the phase difference.

Systems and methods are described for dynamically suppressing non-linear distortion for a device, such as a speakerphone. A device may receive a signal, where the device has non-linear distortion at a predetermined frequency. The received signal may be analyzed to compute a tone strength parameter and a band level. The received signal may be filtered such that a spectrum of the input signal is dynamically limited by reducing suppression of the non-linear distortion when the tone strength parameter is in a lower portion of a predetermined range and increasing suppression of the non-linear distortion when the tone strength parameter is in an upper portion of the predetermined range, the predetermined range of the tone strength parameter corresponding to a loudness range of the device.

A computer-implemented method comprising: determining one or more features of a subject signal; revising one or more control signals on the basis of the one or more features; modifying a level of the subject signals based on the control signals. At least one of the features is determined by: comparing a given one of the subject signals against a boundary signal to produce a corresponding given boundary comparison signal; and summarizing the behavior of the given boundary comparison signal over a time interval.