Abstract: Magic cube speaker with charging stand, charging block, and methods for controlling the same, comprising a magic cube module having a liner with a cavity inside and splicing blocks installed on surface of the liner, a speaker inside the cavity, a charging interface, and a button mounted to the splicing block. The splicing block comprises sound passages formed by a sound inlet port, a sound outlet port, and a middle section between the sound inlet and outlet ports. A first sound transmission channel is formed by the cavity inside the liner, the sound transmission holes, the sound inlet port, the hollow space inside the splicing block, and the sound outlet port, and a second sound transmission channel is formed by the cavity inside the liner, the sound transmission holes, the sound inlet port, the hollow space inside the splicing block, and the plurality of the micro holes.

Abstract: A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

Abstract: A networked speaker system automatically adjusts certain audio speaker system settings based on whether the system is inside a building or has been moved outside as may be sensed by one or capacitors on a speaker. Also, techniques are described for adjusting speaker driver direction based on walls or other barriers in a room or based on the location of a listener.

Abstract: Systems and methods for data communication using audio patterns are described. The methods include encoding a message into an audio pattern, mixing the audio pattern with an audio content to obtain a mixed audio content, wherein the audio pattern is inaudible in the mixed audio content, and transmitting the mixed audio content.

Abstract: Techniques for presence-detection devices to detect movement of a person in an environment by emitting ultrasonic signals using a loudspeaker that is concurrently outputting audible sound. To detect movement by the person, the devices characterize the change in the frequency, or the Doppler shift, of the reflections of the ultrasonic signals off the person caused by the movement of the person. However, when a loudspeaker plays audible sound while emitting the ultrasonic signal, audio signals generated by microphones of the devices include distortions caused by the loudspeaker. These distortions can be interpreted by the presence-detection devices as indicating movement of a person when there is no movement, or as indicating lack of movement when a user is moving. The techniques include processing audio signals to remove distortions to more accurately identify changes in the frequency of the reflections of the ultrasonic signals caused by the movement of the person.

Abstract: An n-phonic energy detection (“NED”) system includes two antenna structures separated by a distance and configured to be placed adjacent one of a pair of human ears. Each of the two antenna structures includes antenna elements. The NED system also includes speakers configured to be placed adjacent one of the pair of human ears. The NED system also includes radio frequency (“RF”) detectors configured to detect RF energy emitted from a source and received by the two antenna structures, and an amplifier that amplifies signals from the RF detectors and outputs the amplified signals to a computer and to the speakers corresponding to the antenna structure to be placed adjacent the same one of the pair of human ears.

Abstract: In accordance with an embodiment, a system includes a digital calibration filter device configured to receive a digital input signal based on a sensor output signal from a sensor, receive a sensor-specific control signal, and perform digital filter processing of the digital input signal to produce a calibrated output signal, wherein the digital filter processing is based on the sensor-specific control signal; and a control device configured to select the sensor-specific control signal from a plurality of sensor-specific control signals based on an ascertained influencing parameter, and provide the sensor-specific control signal to the digital calibration filter device.

Abstract: An interactive control method and device of earphones, an earphone and a storage medium are disclosed. The earphones include a first earphone and a second earphone. The method comprises: acquiring putting-on sequential order information or taking-off sequential order information of the first earphone and/or the second earphone; determining a first functional instruction corresponding to the putting-on sequential order information or a second functional instruction corresponding to the taking-off sequential order information according to a preset correspondence between the putting-on sequential order information or the taking-off sequential order information and functional instructions of a mobile terminal; and sending the first functional instruction or the second functional instruction to the mobile terminal, so that the mobile terminal executes the first functional instruction or the second functional instruction to realize a first interaction function or a second interaction function with the earphones.

Abstract: An analysis system has at least one monitoring assembly that includes at least one microphone to monitor an acoustic field proximate the monitoring assembly. The at least one microphone produces at least one microphone signal responsive to the acoustic field. The analysis system further includes a data storage device configured to buffer the at least one microphone signal and an audio analysis system configured to analyze a content of the data storage device where the audio analysis system is configured to analyze the content of the buffer to process at least a sound into a response or action.

Abstract: An audio signal processing apparatus is provided. The audio signal processing apparatus includes an input terminal receiving an input audio signal, a processor obtaining a difference between a playback loudness of the input audio signal and a desired loudness thereof and correcting a frequency band spectrum of an output audio signal for each of a plurality of frequency bands based on the difference between the playback loudness and the desired loudness of the input audio signal and a relationship between a loudness and a sound pressure for each of the plurality of frequency bands, and an output terminal outputting the output audio signal. The playback loudness is a loudness of the output audio signal when the input audio signal is output without the correction.

Abstract: An information processing device includes a first time-frequency converter configured to perform a time-frequency conversion with respect to an input signal, using a window function having a first width, a second time-frequency converter configured to perform a time-frequency conversion with respect to the input signal, using a second window function having a second width smaller than the first width, and a modification processing unit configured to modify an output of the second time-frequency converter, using a frequency analysis result based on an output of the first time-frequency converter.

Abstract: An equalizer and a method of controlling same are provided. The equalizer includes a memory storing an EQ value set for a plurality of music attributes and storing a general-purpose EQ value; and a processor configured to: obtain an input music signal; calculate a plurality of probability values for the plurality of music attributes by analyzing attributes of the input music signal based on a convolutional neural network; calculate a moderate index between the plurality of probability values; generate an EQ value based on the plurality of probability values and the moderate index; and perform equalizing by applying the generated EQ value to the input music signal.

Abstract: A driving sound library provides driving sounds classified in various themes to a user by analyzing frequency characteristics and temporal characteristics of sound sources classified into categories and determining a chord corresponding to each sound source based on the frequency characteristics and the time characteristics. Modulated sound sources in which each sound source is modulated are generated by applying a chord corresponding to each sound source. The driving sound sources are generated using the sound sources and the modulated sound sources as input data. Pitches of the driving sound sources are then changed based on the driving sound sources and engine sound orders corresponding to an engine RPM of a preset range. Scores for each theme for the driving sounds are generated.

Abstract: A multi-channel and multi-zone audio environment is provided. Various inventions are disclosed that allow playback devices on one or more networks to provide an effective multi-channel and a multi-zone audio environment using timing information. According to one example, timing information is used to coordinate playback devices connected over a low-latency network to provide audio along with a video display. In another example, timing information is used to coordinate playback devices connected over a mesh network to provide audio in one or more zones or zone groups.

Abstract: An Nth 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 ?. An Nth order polynomial associated with the target beampattern may then be generated. A relationship between the nulls of the polynomial and the steering angle ? 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 ?. 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 Nth order and the determined null value may be associated with the null of the polynomial that is of Nth order.

Abstract: Systems and methods of providing improved directional noise suppression in an electronic device implement a technique that specifies a direction or speaker of interest, determines the directions corresponding to speakers not lying in the direction of interest, beam forms the reception pattern of the device microphone array to focus in the direction of interest and suppresses signals from the other directions, creating beam formed reception data. A spatial mask is generated as a function of direction relative to the direction of interest. The spatial mask emphasizes audio reception in the direction of interest and attenuates audio reception in the other directions. The beam formed reception data is then multiplied by the spatial mask to generate an audio signal with directional noise suppression.

Abstract: A noise reduction method and apparatus for an on-board environment, an electronic device and a storage medium are provided, which are applicable to a field of computer technology, and particularly to a field of audio processing. The noise reduction method for an on-board environment includes: receiving an interference signal in the on-board environment and receiving a sound signal in the on-board environment, the interference signal comprising a vibration signal of a vehicle; and performing noise reduction processing on the sound signal in the on-board environment to obtain a noise-reduced signal; wherein, the noise reduction processing comprises cancelling the interference signal from the sound signal in the on-board environment.

Abstract: A speech enhancement method and a speech enhancement system are provided. The speech enhancement method performs two-stage noise suppression by using digital signal processing and neural network approach. The first-stage noise suppression generates artifact signals by reducing stationary noise in the digital audio signals. The second-stage noise suppression performs voice activity detection and further reduces non-stationary noise in the artifact signals. The result of the voice activity detection is fed back to establish or update a noise model used in the first-stage noise suppression.

Abstract: A conversation support device includes: a speaker; a microphone; a noise source acquisition unit that acquires a noise signal indicating noise; a first calculator that calculates a transfer characteristic of a secondary path between the speaker and the microphone; an echo cancellation unit that cancels an echo by using the transfer characteristic of the secondary path; a second calculator that calculates a coefficient of an adaptive filter, based on the transfer characteristic of the secondary path and the noise signal; and an active noise cancellation controller that generates a noise cancelling signal by using the coefficient of the adaptive filter and the noise signal. The noise cancelling signal is for controlling cancellation of the noise.

Abstract: The present disclosure relates to circuitry comprising: amplifier circuitry configured to receive a variable supply voltage, wherein the supply voltage varies according to an output signal of the amplifier circuitry; monitoring circuitry configured to monitor one or more parameters of an output signal of the amplifier circuitry; and processing circuitry configured to receive an indication of the voltage of the variable supply voltage and an indication of the monitored parameters from the monitoring circuitry and to apply a correction to one or more of the monitored parameters to compensate for coupling between the variable supply voltage and the monitoring circuitry.