Abstract: 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.

Abstract: A computer-implemented method for generating anti-noise using an anti-noise generator to suppress noise from a noise source in an environment comprises processing a sound signal, which is representative of ambient sound including noise, anti-noise and propagation noise from the environment, using a deep learning algorithm configured to generate an anti-noise signal to form anti-noise. The deep learning algorithm comprises a convolution layer; after the convolution layer, a series of atrous scaled convolution modules, wherein each of the atrous scaled convolution modules comprises an atrous convolution, a nonlinear activation function after the atrous convolution, and a pointwise convolution after the nonlinear activation function; after the series of atrous scaled convolution modules, a recurrent neural network; and after the recurrent neural network, a plurality of fully connected layers.

Abstract: Methods and systems for actively and adaptively deconstructively interfering with noise and vibration generated by medical devices. A method includes receiving, via user input, information associated with a configuration of the medical device or a physical environment proximate to the medical device. The method includes determining a signature signal for the medical device based at least on the information and sensor data from a sensor of the medical device configured to sense a characteristic of the physical environment proximate to the medical device, the signature signal representing physical attributes of the physical environment; and determining an inverted signal based on the signature signal.

Abstract: An electronic device is provided. The electronic device includes a speaker, a microphone, a processor, and a memory. For example, the electronic device may obtain a first audio signal during a first specified time by using the microphone, may identify reference noise intensity based on the first audio signal, may obtain a second audio signal exceeding the reference noise intensity by using the microphone, may turn off the microphone based on a fact that a third audio signal having the reference noise intensity or less is obtained during a second specified time or longer, and may modulate and output the second audio signal through the speaker while the microphone is turned off.

Abstract: A sound control device mounted in a vehicle and control method include measuring a drive signal input to a speaker, wherein the drive signal is generated in response to an input signal including at least one of a noise control signal and an audio signal, estimating a state of a voice coil of the speaker including at least one of displacement and temperature of the voice coil, based on the drive signal and a model of the speaker, and adjusting the input signal based on the state of the voice coil.

Abstract: An acoustic echo canceller (AEC) uses a simulated echo from an adaptive filter to cancel an actual echo from a real environment. The echo cancellation is optimal when the adaptive filter response accurately matches the environment response. This can be achieved by changing coefficients of the filter in a step-wise fashion until the filter is converged on a configuration that provides optimal results. The convergence of the filter can be negatively affected by interference due to a double talk event or by an event in which the environment is changed. The disclosed AEC is configured to respond to these disruptive events by adjusting a step-size of the adaptive algorithm used to change the coefficients.

Abstract: An ultra slim micro-speaker is provided with a middle gasket between the voice coil and the diaphragm is provided. A distance between a top of a top plate and a top of the voice coil can be equal to a distance between a bottom of the top plate and a bottom of the voice coil, and a winding height has a symmetrical relationship in the magnetic circuit with a BL(x) curve of the magnetic circuit having improved symmetry.

Abstract: The present disclosure relates to a method, a device, and a computer program for controlling and managing a state of a peripheral device in a wireless communication system, and a recording medium therefor. According to an embodiment of the present disclosure, a method for changing an audio stream session-related state may include the steps of: acquiring information relating to a current audio stream session-related state of a second device before a first device in a first audio stream session-related state performs transition of an audio stream session-related state; determining a second audio stream session-related state of the first device at least on the basis of the current audio stream session-related state of the second device; and performing the transition of the audio stream session-related state by the first device on the basis of the second audio stream session-related state.

Abstract: An apparatus includes a receiver and an up-mixer. The receiver is configured to receive a bitstream that includes an encoded mid signal and encoded stereo parameter information. The encoded stereo parameter information represents a first value of a stereo parameter and a second value of the stereo parameter. The first value is associated with a first frequency range. The second value is associated with a second frequency range that is distinct from the first frequency range. The up-mixer is configured to perform an up-mix operation on a frequency-domain decoded mid signal generated from the encoded mid signal. A particular value based on the first value and the second value is applied to the frequency-domain decoded mid signal during the up-mix operation.

Abstract: An active noise cancelling cord contains: a plug including a ground-wire conductor, a live-wire conductor, and a neutral-wire conductor; a socket including a protrusion in which a first receiving orifice, a second receiving orifice and a third receiving orifice are defined; and a noise reduction mode. The body is connected with a plug and a socket, and a shell is located between the plug and the socket. The conductive set includes a ground wire electrically connected with a ground-wire conductor and a first receiving orifice, a live wire electrically connected with a live-wire conductor and a second receiving orifice, a neutral wire electrically connected with a neutral-wire conductor and a third receiving orifice, a plug anti-noise wire electrically connected with a live-wire conductor, and the noise reduction mode, and a socket anti-noise wire electrically connected with the second receiving orifice and the noise reduction mode.

Abstract: An ultra slim micro-speaker is provided with a middle gasket between the voice coil and the diaphragm is provided. A distance between a top of a top plate and a top of the voice coil can be equal to a distance between a bottom of the top plate and a bottom of the voice coil, and a winding height has a symmetrical relationship in the magnetic circuit with a BL(x) curve of the magnetic circuit having improved symmetry.

Abstract: Systems and methods are provided for architectures for a digital class D driver that increase the power efficiency of the class D driver. In particular, systems and methods are provided for a digital class D driver having a feedback analog-to-digital converter (ADC) that can have a latency of 1 cycle or more than 1 cycle. A feedback ADC with a latency of 1 cycle or more is significantly lower power than a low latency feedback ADC. Systems and methods are disclosed for a power efficient digital class D driver architecture that allows for a latency of one or more cycles in the feedback ADC.

Abstract: The present disclosure provides a computer-implemented method for sound masking. The method includes identifying one class among a plurality of classes based on frequency characteristics of a voice of a first occupant in a vehicle, obtaining reference voice data corresponding to the identified class, generating a personalized masking sound for the first occupant by synthesizing the reference voice data with white noise, and controlling an output of the personalized masking sound.

Abstract: A masking sound adjustment method includes obtaining, in each of a plurality of frequency bands, a volume adjustment amount of a masking sound with respect to a volume of a conversation sound to be masked, based on a threshold value corresponding to a target word intelligibility of the conversation sound to be masked; and adjusting a volume of the masking sound in each of the plurality of frequency bands based on the volume adjustment amount.

Abstract: A controlling method including controlling signal processing of the channel assigned to each of the plurality of first physical controllers, based on a parameter indicated by each of the plurality of first physical controllers, enabling a parameter change operation by the plurality of first physical controllers when a channel group set to be enabled by the third physical controller is designated, and disabling the parameter change operation by the plurality of first physical controllers when a channel group set to be disabled by the third physical controller is designated.

Abstract: A device operates to perform acoustic echo cancellation. The device includes a speaker to output a far-end signal at the device, a microphone to receive at least a near-end signal and the far-end signal from the speaker to produce a microphone output, and an AI accelerator operative to perform neural network operations according to a first neural network model and a second neural network model to output an echo-suppressed signal. The device further includes a digital signal processing (DSP) unit. The DSP unit is operative to perform adaptive filtering to remove at least a portion of the far-end signal from the microphone output to generate a filtered near-end signal, and perform Fast Fourier Transform (FFT) and inverse FFT (IFFT) to generate input to the first neural network model and the second neural network model, respectively.

Abstract: An audio signal processing method implemented by an audio system with an audio sensor and a speaker unit includes: measuring, by the audio sensor, acoustic signals reaching the audio sensor, producing a sensor audio signal; retrieving a speaker audio signal corresponding to a speaker acoustic signal from the speaker unit while measuring the acoustic signals reaching the audio sensor to produce the sensor audio signal; converting the speaker and sensor audio signals to speaker and sensor audio spectra; estimating, based on the speaker audio spectrum, an echo audio spectrum of an echo audio signal caused by the speaker acoustic signal in the sensor audio signal; computing, based on the echo audio spectrum and the sensor audio spectrum, echo suppression gains to be applied to the sensor audio spectrum, by using a MMSE-LSA estimator; and applying the echo suppression gains to the sensor audio spectrum.

Abstract: Herein described may be an embodiment of a surround sound headphone device for the production of surround sound which may comprise a first and second headphone assembly which may have acoustic chamber surrounds. The acoustic chamber surrounds may envelope a first and second acoustic chamber assembly. The first and second acoustic chamber assembly may have an acoustic chamber dividing partition, a one or more than one high frequency external port, a one or more than one low frequency external port, a one or more than one low frequency high frequency port, a one or more than one high frequency high frequency port, a one or more than one high frequency auditory source and, a one or more than one low frequency auditory source.

Abstract: Various implementations include methods and related systems for controlling the audio output in a vehicle cabin. In one implementation, a method includes: receiving a first input indicative of a first audio output at a first location in the vehicle cabin; displaying a range of outputs available to a user in a second location in the vehicle cabin; receiving a second input indicative of a second audio output within the range of outputs at the second location; and outputting the first audio output and the second audio output in the vehicle cabin.

Abstract: The present disclosure pertains to an apparatus, which has circuitry configured to control a loud-speaker arrangement including at least one virtual loudspeaker and at least one real loudspeaker to generate at least one virtual sound source, wherein the at least one virtual sound source is generated based on contributions of the at least one virtual loudspeaker and the at least one real loudspeaker, and wherein a soundfield modulation function configured to generate an acoustic impression for a user is used to modulate a soundfield emitted by the virtual sound source, such that the acoustic impression is generated at a predetermined position.