A hearing aid comprises a sensor configured for detecting a focus of an end user on a real sound source, a microphone assembly configured for converting sounds into electrical signals, a speaker configured for converting the electrical signals into sounds, and a control subsystem configured for modifying the direction and/or distance of a greatest sensitivity of the microphone assembly based on detected focus. A virtual image generation system comprises memory storing a three-dimensional scene, a sensor configured for detecting a focus of the end user on a sound source, a speaker configured for conveying sounds to the end user, and a control subsystem configured for causing the speaker to preferentially convey a sound originating from the sound source in response to detection of the focus, and for rendering image frames of the scene, and a display subsystem configured for sequentially displaying the image frames to the end user.

A binaural aid assistance system comprises first and second hearing assistance devices, each being adapted for being located at or in, or fully or partially implanted in the head at one of a left and right ear, respectively, of a user, wherein each of the first and second hearing assistance devices comprises a) a first wireless interface comprising first antenna and transceiver circuitry adapted for establishing a first communication link to the respective other hearing assistance device based on near-field communication; b) a second wireless interface comprising second antenna and transceiver circuitry adapted for establishing a second bi-directional communication link to an auxiliary device based on far-field communication; and wherein the hearing assistance system is configured to transmit an audio signal picked up by a microphone of the hearing assistance device(s) to said auxiliary device.

In a method of audio reproduction in a hearing device, a first external signal is provided, a geometric data set is predetermined for a head shape of a user of the hearing device, and a first position is predetermined for a first virtual speaker. A propagation of the first external signal from the first virtual speaker to a first local unit of the hearing device is simulated based on the geometric data set for the head shape of the user and on the first position and a first virtual spatial signal is generated in the process. A first reproduction signal is generated from the first virtual spatial signal, and a first output transducer reproduces the first reproduction signal in the first local unit of the hearing device.

Systems and methods for processing an audio signal are provided for replay on an audio device. An audio signal is spectrally decomposed into a plurality of subband signals using and pass filters. Each of the subband signals are provided to a respective modulator and subsequently, from the modulator output, provided to a respective first processing path that includes a first dynamic range compressor, DRC. Each subband signal is feedforward compressed by the respective first DRC to obtain a feedforward-compressed subband signal, wherein the first DRC is slowed relative to an instantaneous DRC. Subsequently, each feedforward-compressed subband signal is provided to a second processing path that includes a second DRC, wherein the feedforward-compressed subband signal is compressed by the respective second DRC and outputted to the respective modulator. Modulation of the subband signals is then performed in dependence on the output of the second processing path. Finally, the feedforward-compressed subband signals are recombined.

A method for adjusting the clarity of an audio output in a changing environment, including: receiving a content signal; applying a customized gain to the content signal; and outputting the content signal with the customized gain to at least one speaker for transduction to an acoustic signal, wherein the customized gain is applied on a per frequency bin basis such that frequencies of a lesser magnitude are enhanced with respect to frequencies of a greater magnitude and an intelligibility of the acoustic signal is set approximately at a desired level, wherein the customized gain is determined according to at least one of a gain applied to the content signal, a bandwidth of the content signal, and a content type encoded by the content signal.

In customized audio attenuation a computer system generates audible sounds in one or more frequency ranges from electronic signals. An audiogram for a listener is inferred from the listener's response to the audible sounds and an attenuation profile is determined from the audiogram. The attenuation profile includes an attenuation level for each of the one or more frequency ranges. Each attenuation level is inversely related to the listener's sensitivity to hearing sounds in the corresponding frequency range. Subsequent signals or data corresponding to subsequent audible sounds in the one or more frequency ranges are generated. The attenuation profile is applied to the subsequent signals to generate attenuated signals and the attenuated signals are transmitted to an audio transducer.

At least one exemplary embodiment is directed to a method of determining hearing loss using audio cues whose spectrum is orthogonal to each other to identify a wearer's hearing loss, and using the information to adjust audio to normal listening levels.

Disclosed herein are systems and methods for a sound enhancement system coupled to a mobile device. In one aspect, the present disclosure includes the system having first bidirectional communication link configured to receive an audio signal, transmit an audio signal, and receive configuration parameters. The system also includes a memory configured to store the configuration parameters. Additionally, the system has an audio input configured to receive ambient audio. Yet further, the is configured to operate in one of two operation modes. In the first mode, the system is configured to receive an audio signal from the bidirectional communication link and output an audio signal based on the received audio signal by way of the audio output pathway, and in the second mode, the system is configured to process the ambient audio signal by the audio processor based on stored configuration parameters and create an output audio signal.

A sound outputting apparatus which includes a communicator communicate with an electronic apparatus and receive first audio data, an output module to output the first audio data received, and to output second audio data, which is modified data of the first audio data, a sensor to detect brainwave data of a user, and a processor to control so that the brainwave data of the user detected through the sensor is transmitted to the electronic apparatus through the communicator, and so that the second audio data is received from the electronic apparatus.

A portable audio system with an integral hearing test is disclosed. The device includes a plurality of filter circuits. A processor applies a respective audio frequency to each filter circuit in a test mode to determine a respective gain based on a user input and applies the respective gain to each filter circuit in a normal mode. A switch circuit selects an audio signal from a plurality of sources in the normal mode. An analog-to-digital converter converts the selected audio signal to a digital signal and applies the digital signal to the plurality of filter circuits. A sum circuit receives a digital output signal from each of the plurality of filter circuits and produces a combined signal. A digital-to-analog converter converts the combined signal to an analog output signal.