A method for designing a line array loudspeaker arrangement comprises providing a loudspeaker arrangement based on design start parameters and including at least a vertical front array and measuring the frequency responses of the loudspeaker arrangement with bypassed or omitted electronic filters at predefined horizontal angle increments. The method further comprises computing combined beam forming and crossover filter frequency responses for the vertical front array based on the measured frequency responses of the loudspeaker arrangement and first target frequency responses at various frequency points and various positions; and computing combined equalizing and crossover filter frequency responses for the vertical front array based on second target frequency responses, the combined equalizing and crossover filter frequency responses being configured to obtain acoustic linear phase responses of the loudspeaker arrangement.

A suspension system is provided with a frame to mount to a side surface of a speaker cabinet with a plurality of apertures formed therethrough. The suspension system is also provided with a lever arm with a proximal end and a distal end that are both adapted to connect to an upper support. The lever arm also has an intermediate portion that is pivotally connected to the frame about a pivot axis with a plurality of arcuate slots formed through. Each arcuate slot is formed with an end stop that corresponds to a splay angle of the speaker cabinet relative to the upper support, and each slot aligns with one of the apertures to receive a pin. Whereby suspending the speaker cabinet from the upper support pivots the lever arm in a first direction until the pin engages the end stop.

Noise is suppressed from a microphone array by estimating a noise field isotropy. In some examples audio is received from a plurality of microphones. A power spectral density of a beamformer output is determined and a power spectral density of microphone noise differences is determined. A noise power spectral density is determined using a transfer function and the noise power spectral density is applied to the beamformer output power spectral density to produce a power spectral density output of the received audio with reduced noise.

Provided is an information processing device that performs processing on a content. An information processing device is provided with an estimation unit that estimates sounding coordinates at which a sound image is generated on the basis of a video stream and an audio stream, a video output control unit that controls an output of the video stream, and an audio output control unit that controls an output of the audio stream so as to generate the sound image at the sounding coordinates. A discrimination unit that discriminates a gazing point of a user who views video and audio is further provided, in which the estimation unit estimates the sounding coordinates at which the sound image of the object gazed by the user is generated on the basis of a discrimination result.

A videoconferencing system includes a camera acquiring image data and a microphone array acquiring audio data. Image data is used in conjunction with sound source localization (SSL) data to locate a talker depicted in the image data. SSL processes the audio data and determines SSL pan angle values indicative of an estimated direction of a sound. Columns of pixels in an image are associated with bins. A bin count is incremented for each SSL pan angle value of the audio data that falls within a given bin. A bounding box in the image data is determined that encompasses a face depicted in the image data. A range of pixels is determined for the bounding box, such as extending from a leftmost column to a rightmost column. The bin with the highest bin count that also overlaps a range of pixels for a bounding box is deemed to contain the talker.

Systems, apparatuses, and methods are described for controlling source tracking and delaying beamforming in a microphone array system. A source tracker may continuously determine a direction of an audio source. A source tracker controller may pause the source tracking of the source tracker if a user may continue to speak to the system. The source tracker controller may resume the source tracking of the source tracker if the user may cease to speak to the system, or when one or more pause durations have been reached.

The present disclosure discloses an in-vehicle independent sound zone control method, a system and a related device, applied to a vehicle. The method includes the following steps: presetting a control area and a non-control area; arranging a speaker array behind a front seat of the vehicle for generating a first acoustic response, and arranging a headrest speaker at a headrest on a rear seat of the vehicle for generating a second acoustic response; fitting a virtual target speaker, wherein the virtual target speaker is configured to generate a target acoustic response within the control area; and controlling a sound quality of the in-vehicle independent sound zone through an audio algorithm processing on the target acoustic response, the first acoustic response and the second acoustic response.

A target speech signal extraction method for robust speech recognition includes: initializing a steering vector for a target speech source and an adaptive vector, setting a real output channel of the target speech source as an output by the adaptive vector, initializing adaptive vectors for a noise and setting a dummy channel as an output by the adaptive vectors for the noise; setting a cost function for minimizing dependency between a real output for the target speech source and a dummy output for the noise; setting an auxiliary function to the cost function, and updating the adaptive vector for the target speech source and the adaptive vectors for the noise by using the auxiliary function and the steering vector; estimating the target speech signal by using the adaptive vector thereby extracting the target speech signal from the input signals; and updating the steering vector for the target speech source.

Apparatus and method provided herein are directed to a linear differential directional microphone array (LDDMA), which takes into account the directionality of the array elements. The LDDMA may be designed by generating a steering vector for a linear array (LA) having preselected parameters including parameters δ, p, θ, N, and M, generating a constraint matrix based on the steering vector, reformulating the constraint matrix based on a microphone response matrix and a steering matrix, obtaining a beamformer by applying a minimum norm solution in terms of the constraint matrix, verifying a desired characteristic of the LA by calculating the beamformer for a desired direction, and constructing the LA based on the preselected parameters and the beamformer.

The disclosure relates to a soundbar and a method for automatic surround pairing and calibration of a surround sound system. The soundbar includes two built-in microphones on the left and right respectively, which can be used for determining relative positions of left and right surround speakers. When the relative positions of the left and right speakers are not correct, configurations of left and right surround channels can be automatically swapped with each other without manually swapping physical positions of the surround speakers by a user. In addition, latencies including a latency of each channel of a main system may also be calibrated, and magnitude compensation may be achieved by calculating a filter compensation coefficient of each line and merging it into an original filter. The automatic surround pairing and calibration of the surround sound system may be one-click completed automatically by the user by pressing a start button.