Abstract: Disclosed are a method, an apparatus and an electronic device for controlling audio playback of multiple loudspeakers, wherein the method comprises: determining the location information of each speaker and voice signals issued by each speaker; determining the area where each speaker is located according to the location information of each speaker; determining voice instruction corresponding to each voice signal; and controlling the multiple loudspeakers to play the audio indicated by the corresponding voice instruction respectively for the area where the speaker of each voice instruction is located. According to the method and apparatus and/or the electronic device in an embodiment of the present disclosure, different audios can be played for different areas in a preset space.

Abstract: A method for automated setting of digital processing parameters intended to be applied to digital signals before broadcast by at least one loudspeaker placed in an environment. The method includes a step of determining a set of frequency responses of the environment, an equalization step applied for each subset of at least one loudspeaker and a second equalization step applied to at least one subset of at least one loudspeaker. A device for the implementation of the method is also provided.

Abstract: A method implemented in an audio synthesis device for synthesizing an audio signal is provided. The method includes determining a first plurality of harmonics based on a sinusoidal oscillator, at least two of the first plurality of harmonics being calculated in parallel, scaling the first plurality of harmonics according to a scaling parameter, determining a first sum of the first plurality of scaled harmonics to generate a first sample of the plurality of samples, determining a second plurality of harmonics based on the sinusoidal oscillator, at least two of the second plurality of harmonics being calculated in parallel, scaling the second plurality of harmonics according to the scaling parameter, determining a second sum of the second plurality of scaled harmonics to generate a second sample of the plurality of samples, and causing playback, on the speaker, of at least the first sample and the second sample.

Abstract: A system provides for psychoacoustic frequency range extension. The system generates quadrature components from an audio channel, and generates rotated spectral quadrature components by applying a forward transformation that rotates a spectrum of the quadrature components from a standard basis to a rotated basis. In the rotated basis, the system isolates components of the rotated spectral quadrature components at target frequencies, and generates weighted phase-coherent harmonic spectral quadrature components by applying a nonlinearity to the isolated components having a dependence on scale that is subject to constraints. The circuitry generates a harmonic spectral component by applying an inverse transformation that rotates a spectrum of the weighted phase-coherent harmonic spectral quadrature components from the rotated basis to the standard basis.

Abstract: A method, a system, and a device for transmitting data through an earphone and a computer storage medium are provided. A first data type of to-be-transmitted data is acquired, a second data type set for an FIR filter coefficient is acquired, and a sum of the number of decimal places indicated by the first data type and the number of decimal places indicated by the second data type is determined. A data type which indicates a number of decimal places that is at least two orders more than the sum is determined among preset data types as a third data type of an FIR filtering result, and FIR filtering is performed on the to-be-transmitted data to obtain a target filtering result and transmit the target filtering result.

Abstract: Disclosed are systems and methods for performing adaptive equalization operations to reduce variations in the frequency response of audio signals at the eardrum of a wearer of an earphone. The assumption that the frequency response of the error microphone matches the frequency response at the eardrum may not be true due to signal leakage effects and the shape of the ear canal. Techniques are disclosed for the adaptive equalization operations to perform a calibration algorithm to compensate for the effects of the signal leakage and the shape of the ear canal. The earphone may estimate the transfer function from the speaker to the error microphone and may estimate the load impedance from the earphone based on a calibrated relationship between the transfer function and the load impedance. The adaptive equalization may use a calibration algorithm to adjust the frequency response at the error microphone to match that at the eardrum.

Abstract: A spatial normalization unit generates a normalized spectrum by normalizing an orientation component of a microphone array for a target direction included in a spectrum of an acoustic signal acquired from each of a plurality of microphones forming the microphone array into an orientation component for a predetermined standard direction. A mask function estimating unit determines a mask function used for extracting a component of a target sound source arriving in the target direction on the basis of the normalized spectrum using a machine learning model. A mask processing unit estimates the component of the target sound source installed in the target direction by applying the mask function to the acoustic signal.

Abstract: Example techniques relate to audio generation in a media playback system. Based on one or more first functions and first characteristics of an area, the system may generate first audio that includes a first audio signal and a second audio signal. The system provides the first audio signal to at least one first audio driver and the second audio signal to at least one second audio driver, thereby causing a first playback device and a second playback device to play back the first audio synchronously. The system receives second characteristics of the area and based on one or more second functions and the second characteristics, generates second audio comprising a third audio signal and a fourth audio signal. The system provides the third audio signal to the at least one first audio driver and the fourth audio signal to the at least one second audio driver.

Abstract: A first subwoofer may be configured to output multimedia content in synchrony with at least one other playback device and a second subwoofer. The first subwoofer may, based on a received indication of an acoustic characteristic of the at least one other playback device, determine a crossover frequency of (i) the first subwoofer and the second subwoofer and (ii) the at least one other playback device. After determining the crossover frequency, the first subwoofer may output a first tone set and a second tone set in synchrony with the second subwoofer and the at least one other playback device, and after outputting the first tone set and the second tone set, receive, from a controller device, an indication of a selected one of the first tone set or the second tone set. Based on the selected tone set, the first subwoofer may adjust a phase setting of the first subwoofer.

Abstract: A processor-based method of reducing loudspeaker-induced harmonic distortion in sound generated by a loudspeaker responsive to a first audio signal, the method including: attenuating a first band in the first audio signal by an attenuation gain, the first band including a fundamental frequency, wherein the attenuating dynamically adjusts the attenuation gain in accordance with, at least, a ratio between: i) an energy of the first band in the first audio signal; and ii) an energy of a second band in the first audio signal, the second band including one or more frequencies that is an integer multiple of the fundamental frequency.

Abstract: An example method of operation may include determining a frequency response to a measured chirp signal detected from one or more speakers, determining an average value of the frequency response based on a high limit value and a low limit value, subtracting a measured response from a target response, and the target response is based on one or more filter frequencies; determining a frequency limited target filter with audible parameters based on the subtraction, and applying an infinite impulse response (IIR) biquad filter based on an area defined by the frequency limited target filter to equalize the frequency response of the one or more speakers.

Abstract: Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reflection of the input audio signal in a surface of the virtual environment.

Abstract: Provided is a vibration control apparatus that acquires content information about content including at least either one of video and audio to be presented to a user, and vibrates a vibration mechanism in such a manner that the user feels a pseudo force sensation defined based on the content information while the content is being presented.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for dynamic volume adjustment via audio classification.

Abstract: Endfire linear array microphone systems and methods with consistent directionality and performance at different frequency ranges are provided. The endfire linear array microphone includes a delay and sum beamformer and a differential beamformer. The delay and sum beamformer may produce pickup patterns with good directionality at higher frequency ranges, but cause the pickup patterns to become more omnidirectional at lower frequencies. The differential beamformer may produce pickup patterns with good directionality at lower frequencies. By combining the delay and sum beamformer and differential beamformer within the linear array microphone, the overall directionality of the linear array microphone may be maintained at different frequency ranges while using the same microphone elements.

Abstract: Methods and systems for identifying abnormal sounds in a particular environment. A normal audio stream obtained in the absence of abnormal sounds may be used as a baseline for subsequently processing an incoming audio stream with a processor to determine whether the incoming audio stream from the microphone in the particular environment includes an abnormal audio event for the particular environment. When it is determined that the incoming audio stream includes an abnormal audio event for the particular environment an electronic database may be accessed to determine a location of the abnormal audio event in the particular environment. A video camera with a field of view that includes the location of the abnormal audio event in the particular environment may be identified and the video stream from the identified video camera retrieved and displayed.

Abstract: A hearing system to activate an auditory system using cerebrospinal fluids includes at least one processor configured to receive an audio signal captured using a sound sensor (e.g., a microphone), extract temporal and spectral features from the audio signal, and create modulated ultrasound signals in a range of 20 Hz to 20 kHz with ultrasound carrier frequencies in the range of 50 kHz to 4 MHz, which are ultrasound frequencies that are well-suited to reach the cerebrospinal fluids (e.g., can pass across the skull/bones to reach the cerebrospinal fluids). The system further includes at least one ultrasonic transducer which receives the modulated signal and delivers the modulated signal to the body and activates the auditory system via vibration of cerebrospinal fluids that vibrate cochlear fluids, bypassing the normal conductive pathway that uses middle ear bones and minimizing bone conduction and distortion through the skull.

Abstract: A signal processing device includes: an obtainer that obtains sound data; a pitch controller that controls pitch of the sound data obtained to increase the pitch by a factor of n (a real number greater than 1); an extractor that extracts frequency components of at least 20 kHz which are included in the sound data whose pitch is controlled; an adder that adds the frequency components extracted to the sound data obtained; and an outputter that outputs sound data to which the frequency components extracted are added.

Abstract: The present technology relates to a signal processing apparatus and method, and a program that make it possible to reduce an arithmetic operation amount. The signal processing apparatus performs, on the basis of audio object mute information indicative of whether or not a signal of an audio object is a mute signal, at least either one of a decoding process or a rendering process of an object signal of the audio object. The present technology can be applied to a signal processing apparatus.

Abstract: An audio signal processing method includes receiving an audio signal corresponding to a voice of a talker, obtaining an image of the talker, estimating position information of the talker using the image of the talker, generating, according to the estimated position information, a correction filter configured to compensate for an attenuation of the voice of the talker, performing filter processing on the audio signal using the generated correction filter, and outputting the audio signal on which the filter processing has been performed.