There is provided a noise reduction headset having multi-microphone and a noise reduction method. One embodiment of the headset includes a headband and headset bodies respectively connected to both ends of the headband, wherein the headset further includes a microprocessor and at least three microphones disposed on an outer surface of the headset body; the microprocessor is configured to select the microphone closest to a target sound source as a main microphone according to a comparison of sound signals picked up by respective microphones, select the microphone farthest from the main microphone as an auxiliary microphone according to a preset relative position of respective microphones, and execute a noise reduction algorithm by using a first sound signal picked up by the main microphone and a second sound signal picked up by the auxiliary microphone as inputs of the noise reduction algorithm so as to realize noise reduction. According to the embodiment, the noise reduction performance can be ensured under the condition that the user does not wear the headset normally.

A beamforming microphone array may be integrated into a wall or ceiling tile as a single unit. The beamforming microphone array includes a plurality of microphones that picks up audio input signals. In addition, the wall or ceiling tile may include an acoustically transparent outer surface on the front side of the tile, and the beamforming microphone array picks up the audio input signals through the outer surface of the tile. The beamforming microphone array may be coupled to the tile as a single unit and may be integrated into the back side of the tile.

The present disclosure relates to an acoustic output apparatus. The acoustic output apparatus may include an earphone core including at least one acoustic driver for outputting sound though one or more sound guiding holes set on the acoustic output apparatus, a controller configured to cause the at least one acoustic driver to output sound, a power source assembly configured to provide electrical power to the earphone core, the one or more sensors, and the controller, and an interactive control component configured to allow an interaction between a user and the acoustic output apparatus.

An audio processing system includes a microphone array, a speech detection system, and a neural network noise reduction module. The microphone array includes at least two microphones and provides an audio signal from an environment surrounding the microphone array. The speech detection system receives the audio signal, and processes the audio signal to a) detect that a first user is speaking, b) determine a first direction relative to the audio array when the first user is located at a first location within the environment, and c) provide beamforming processing on the audio signal in the first direction, and to provide a processed audio signal based upon the beamforming processing. The neural network noise reduction module reduces noise in the processed audio signal.

A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are configured to apply one adaptive network, based on a constraint that includes preservation of a spatial direction of one or more audio sources in the soundfield at the different time segments, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint. The one or more processors are also configured to apply an additional adaptive network.

A sound pickup device includes microphone elements arranged three-dimensionally in a distributed manner. A total number of effective microphone pairs is greater than a total number of the microphone elements, the effective microphone pairs each being a combination of two microphone elements having a distance less than a distance D between each other. The distance D is represented by D=c/2f, where f represents a frequency of a target sound obtained from each of the microphone elements and c represents a velocity of the target sound. Any one of straight lines each of which connects the two microphone elements of a different one of the effective microphone pairs is not parallel to any other of the straight lines.

Method of performing acoustic zooming starts with microphones capturing acoustic signals associated with video content. Beamformers generate beamformer signals using the acoustic signals. Beamformer signals correspond respectively to tiles of video content. Each of the beamformers is respectively directed to a center of each of the tiles. Target enhanced signal is generated using beamformer signals. Target enhanced signal is associated with a zoom area of video content. Target enhanced signal is generated by identifying the tiles respectively having at least portions that are included in the zoom area, selecting beamformer signals corresponding to identified tiles, and combining selected beamformer signals to generate target enhanced signal. Combining selected beamformer signals may include determining proportions for each of the identified tiles in relation to the zoom area and combining selected beamformer signals based on the proportions to generate the target enhanced signal. Other embodiments are described herein.

A communication system may include, in an example, a first computing device communicatively coupled, via a network, to at least a second computing device maintained at a geographically distinct location than the first computing device; the first computing device including: an array of audio output devices and a processor to receive transmitted speech data and metadata describing an estimated direction of arrival (DOA) of speech from a plurality of speakers at an array of microphones at the second computing device and render audio at the array of audio output devices associated with the first computing device by eliminating spatial collision during rendering; said spatial collision arising due to the low angular separation of the estimated DOA of a plurality of speakers.

A sound generation apparatus includes sound collection means configured to collect a sound of a sound source in a space, image capture means configured to capture an image of the sound source, estimation means configured to estimate an attribute of the sound source from the image captured by the image capture means, sound generation means configured to obtain an acoustic characteristic of a target sound included in the sound collected by the sound collection means and to generate multiple masking sounds on the basis of the acoustic characteristic and the attribute of the sound source estimated by the estimation means, display means configured to display the attribute of the sound source estimated by the estimation means, sound selection means configured to receive selection of a masking sound from the masking sounds generated by the sound generation means, and sound output means configured to output the selected masking sound.

A system includes first, second, and third microphones configured to receive sound waves from a source of the sound waves. The system includes a memory configured to store first, second, and third phase difference maps for the first and second microphones, the second and third microphones, and the third and first microphones. The system includes a processor configured to measure first, second, and third phase differences between the sound waves received from the source by the first and second microphones, the second and third microphones, and the third and first microphones; receive the first, second, and third phase difference maps from the memory; and identify a location of the source of the sound waves based on the first, second, and third phase differences and the first, second, and third phase difference maps for the first and second microphones, the second and third microphones, and the third and first microphones.