An N.sup.th order linear differential microphone array (LDMA) with a steerable beamformer may be constructed by specifying a target beampattern for the LDMA at a steering angle θ. An N.sup.th order polynomial associated with the target beampattern may then be generated. A relationship between the nulls of the polynomial and the steering angle θ is determined and then a value of one of the nulls is determined based on N−1 assigned values for the other nulls and the determined relationship between the nulls of the polynomial and the steering angle θ. The steerable beamformer may be generated based on the determined null value and the N−1 assigned null values. The N−1 assigned null values may be associated with the N−1 nulls of the polynomial that are of less than N.sup.th order and the determined null value may be associated with the null of the polynomial that is of N.sup.th order.

Techniques are described for assisting near distance communications. A first device comprising a receiver, a sensor and a processor may be configured to perform the assisted near distance communication techniques. The receiver may receive, from a second device located within a conversational distance to the first device, monophonic audio data representative of the near distance communication. The sensor may generate a sensor signal representative of spatial information of the near distance communication. The processor may render, based on the spatial information and the monophonic audio data, multi-dimensional audio data in which the near distance communication originates in a soundfield from a location of the second device relative to the first device. The processor may next output the multi-dimensional audio data to a transducer so as to reproduce the near distance communication in multiple dimensions.

A speech recognition system that automatically sets the volume of output audio based on a sound intensity of a command spoken by a user to adjust the output volume. The system can compensate for variation in the intensity of the captured speech command based on the distance between the speaker and the audio capture device, the pitch of the spoken command and the acoustic profile of the system, and the relative intensity of ambient noise.

Method to perform dynamic beamforming to reduce SNR in signals captured by head-wearable apparatus starts with microphones generating acoustic signals. Microphones are coupled to first stem of the apparatus and to second stem of the apparatus. First and second beamformers generate first and second beamformer signals, respectively. Noise suppressor attenuates noise content from the first beamformer signal and the second beamformer signal. Noise content from first beamformer signal are acoustic signals not collocated in second beamformer signal and noise content from second beamformer signal are acoustic signals not collocated in first beamformer signal. Speech enhancer generates clean signal comprising speech content from first noise-suppressed signal and second noise-suppressed signal. Speech content are acoustic signals collocated in first beamformer signal and second beamformer signal.

Disclosed herein are system, apparatus, article of manufacture, method and/or computer program product embodiments, and/or combinations and sub-combinations thereof, for identifying electronic devices in a room using a spread code. In some embodiments, a first electronic device receives a spread spectrum signal from a second electronic device over an audio data channel. The first electronic device determines a time of receipt of the spread spectrum signal based on despreading. The first electronic device calculates a distance between the first electronic device and the second electronic device based on the time of receipt and a time of transmission. The first electronic device determines the second electronic device is not in the room with the first electronic device based on the calculated distance.

There are included a first speaker processing step in which a first speaker 2 produces sound based on a first filtered signal, a gain multiplication step in which a gain multiplication unit 4 generates a gain multiplied signal by multiplying a second filtered signal by a gain, a second speaker processing step in which a second speaker 5 produces sound based on the gain multiplied signal, and a gain control step in which a gain control unit 7 controls the gain such that a root mean square of a collected sound signal collected by a microphone 6 installed at a mute position, which is a position at which the sound produced by the first speaker and the sound produced by the second speaker are to be muted, is relatively small.

A system with a plurality of microphones positioned at different locations, and a modification system in communication with the microphones. The modification system is configured to derive a plurality of audio signals from the plurality of microphones, compute a confidence score for each derived audio signal, and based on the computed confidence scores, use one derived audio signal to modify another audio signal.

Example techniques may involve performing aspects of a spatial calibration. An example implementation may include detecting a trigger condition that initiates calibration of a media playback system including multiple audio drivers that form multiple sound axes, each sound axis corresponding to a respective channel of multi-channel audio content The implementation may also include causing the multiple audio drivers to emit calibration audio that is divided into constituent frames, the multiple sound axes emitting calibration audio during respective slots of each constituent frame. The implementation may further include recording the emitted calibration audio. The implementation may include causing delays for each sound axis of the multiple sound axes to be determined, the determined delay for each sound axis based on the slots of recorded calibration audio corresponding to the sound axes and causing the multiple sound axes to be calibrated.

A sound processing device is provided with a correction unit that corrects a sound pickup signal. The sound pickup signal is obtained by picking up a sound with a microphone array. The correction unit corrects the sound pickup signal based on directional information that indicates a direction of the microphone array in spherical coordinates, during the picking up of the sound.

Eyewear providing a visual indicator to a hearing-impaired user indicating a direction of arrival of a sound relative to the eyewear to help the user obtain greater awareness of the surrounding environment. An eyewear optical assembly includes an image display displaying the visual indicator discernable by the user and corresponding to the direction of arrival of the sound, even when a sound source is not viewable through the optical assembly. The image display also displays an image indicative of the sound source. A front portion of an eyewear frame includes a light array, where one or more lights of the light array is illuminated to indicate the direction of arrival and intensity of the sound source. An array of vibrating devices may also be used to indicate the direction of arrival and intensity of the sound source.