Methods and systems for signal processing an audio signal in a hearing device to detect when feedback path change occurs and controlling an adaptive feedback canceler to remove the feedback are provided. The hearing device includes a receiver and a microphone. An exemplary method includes detecting whether a tonal signal is caused by a feedback path change by estimating a product of a subband error signal and a subband output signal generated in response to a subband audio input signal; estimating a fast metric based on the estimated product and estimating a slow metric; and applying or maintaining an adaptation rate to the adaptive feedback canceler of the hearing device, wherein the adaptation rate applied or maintained is selected based upon a value of the difference between the fast and slow metrics compared to a threshold value.

A method and system or device such as a hearing aid are provided for processing audio signals. In accordance with the method, an audio signal is received and divided into a plurality of frequency sub-bands. For each of the frequency sub-band signals, the signal is further divided into overlapping temporal frames. Each of the temporal frames are windowed. Frequency warping is performed on each of the windowed frames. Overlap-and-add is performed on the frequency warped frames. The frequency warped sub-bands are combined into a full band to provide a frequency warped signal.

A hearing device, e.g. a hearing aid, configured to be worn by a user, comprises a) two or more input transducers (e.g. microphones) wherein said two or more input transducers during use of the hearing device are arranged with a distance between them; b) a directional system comprising a directional algorithm configured to provide a directional pattern in dependence of said distance. The hearing device is configured to estimate a current value of said distance, or an equivalent acoustic delay, or beamformer weights of said directional system, thereby the directional performance can be optimized to the individual user.

An ear-worn electronic system comprise a plurality of discrete devices configured for deployment at one ear of a wearer. The system comprises a first device configured for placement at least partially within an ear canal of the wearer. The first device comprises at least a first processor, one or more first microphones, a first speaker, a first rechargeable battery, and first charging circuitry. The system comprises a second device configured for placement at the wearer's ear proximal of the first device in an outer ear direction. The second device comprises at least a second processor, a second rechargeable battery, and second charging circuitry. The first device is configured to operate autonomously when operating in an independent mode relative to the second device. The first and second devices are configured to operate cooperatively when operating in a communicatively coupled mode.

A hearing aid configured to be worn at, and/or in, an ear of a user, comprises a forward path for processing sound from the environment of the user. The forward path comprises a) at least one first microphone providing at least one first electric input signal representing said sound as received at the respective at least one first microphones, said at least one first microphone being located away from a first ear canal of the user, b) an audio signal processor for processing said at least one first electric input signal, or a signal or signals originating therefrom, and for providing a processed signal, c) an output transducer for providing stimuli perceivable as sound to the user in dependence of said processed signal, and d) at least one second microphone connected to said audio signal processor, the at least one second microphone being configured to provide at least one second electric input signal representing said sound as received at the at least one second microphone, the at least one second microphone being located at or in said first ear canal of the user, and e) a feature extractor for extracting acoustic characteristics of said ear of the user from said at least one second electric input signal, or a signal originating therefrom. The hearing aid is configured to include said acoustic characteristics in the processed signal. The invention may e.g. provide improved sound localization in hearing aids.

An audio system for artificial reality systems is configured to reduce feedback and mitigate entrainment in audio content presented to the user. The audio system utilizes adaptive feedback cancellation processes to reduce feedback. The audio system processes reference signals in a hybrid of the time domain and the frequency domain to achieve rapid convergence with decreased entrainment. The audio system may pre-whiten reference signals and speaker signals and separate the pre-whitened signals into frequency bands using a filter bank. The audio system uses a state space algorithm to generate adaptive filters in each frequency band. The adaptive filters may be applied to the speaker signal to generate a feedback cancelation signal which may be subtracted from the reference signal to decrease feedback.

A personalized sound management system for an acoustic space includes at least one transducer, a data communication system, one or more processors operatively coupled to the data communication system and the at least one transducer, and a medium coupled to the one or more processors. The processors access a database of sonic signatures and display a plurality of personalized sound management applications that perform at least one or more tasks among identifying a sonic signature, calculating a sound pressure level, storing metadata related to a sonic signature, monitoring sound pressure level dosage levels, switching to an ear canal microphone in a noisy environment, recording a user's voice, storing the user's voice in a memory of an earpiece device, or storing the user's voice in a memory of a server system, or converting received text received in texts or emails to voice using text to speech conversion. Other embodiments are disclosed.

Methods and systems for signal processing an audio signal in a hearing device to detect when feedback path change occurs and controlling an adaptive feedback canceler to remove the feedback are provided. The hearing device includes a receiver and a microphone. An exemplary method includes detecting whether a tonal signal is caused by a feedback path change by estimating a product of a subband error signal and a subband output signal generated in response to a subband audio input signal; estimating a fast metric based on the estimated product and estimating a slow metric; and applying or maintaining an adaptation rate to the adaptive feedback canceler of the hearing device, wherein the adaptation rate applied or maintained is selected based upon a value of the difference between the fast and slow metrics compared to a threshold value.

Systems, devices, and methods for communication include an ear canal microphone configured for placement in the ear canal to detect high frequency sound localization cues. An external microphone positioned away from the ear canal can detect low frequency sound, such that feedback can be substantially reduced. The canal microphone and the external microphone are coupled to a transducer, such that the user perceives sound from the external microphone and the canal microphone with high frequency localization cues and decreased feedback. Wireless circuitry can be configured to connect to many devices with a wireless protocol, such that the user can receive and transmit audio signals. A bone conduction sensor can detect near-end speech of the user for transmission with the wireless circuitry in a noisy environment. Noise cancellation of background sounds near the user can be provided.

A hearing aid includes a) at least one input unit for providing a time-frequency representation Y(k,n) of an electric input signal representing sound consisting of target speech and noise signal components, where k and n are frequency band and time frame indices, respectively, b) a noise reduction system configured to b1) determine an a posteriori signal to noise ratio estimate γ(k,n) of the electric input signal, and to b2) determine an a priori signal to noise signal ratio estimate ζ(k,n) of the electric input signal from the a posteriori signal to noise ratio estimate γ(k,n) based on a recursive algorithm providing non-linear smoothing. The a posteriori signal to noise ratio estimate of said electric input signal is provided as a mixture of first and second different a posteriori signal to noise ratio estimates. The invention may be used in audio processing devices, such as hearing aids, headsets, etc.