Abstract: The present disclosure relates to a bi-directional VR (virtual reality) system and, more particularly, to a bi-directional VR system using a user image and a user voice extracted by a 3D depth camera, and media processed by synthesizing an interactive content image and sound. A bi-directional VR system may include two or more terminals; and a mixing server configured to transmit mixed data generated by processing signals transmitted from the terminals to the respective terminals. At least one main terminal of the two or more terminals may include a 3D depth camera; a main body unit provided with a processor and a memory, and loaded with an operation program; at least two sound devices connected to the main body unit to be controlled by the main body unit, and provided with an audio output terminal and a microphone terminal; and at least two display units connected to the main body unit to be controlled by the main body unit.

Abstract: A loudspeaker is disclosed. The loudspeaker includes a frame; a vibration system accommodated in the frame and comprising a vibrating diaphragm and a voice coil for driving the vibrating diaphragm fixed with the frame; a magnetic circuit system driving the vibration system; a light source assembled with the voice coil; a photosensitive sensor located in the yoke and within the irradiation scope of the light source. The magnetic circuit system includes a yoke positioned by the frame and a magnet portion supported by the yoke.

Abstract: The present disclosure is directed to systems, apparatuses, and methods for wirelessly delivering multi-channel audio over a packet based network with tight synchronization, high fidelity, and/or low delay as described above. The systems can include a source device that provides multi-channel audio to a distributor device, which wirelessly distributes the multi-channel audio over the packet based network to audio rendering devices, referred to as “sink” devices. The distributor device and the sink devices each include a playback clock that is used to read audio samples of the multi-channel audio from a local memory and drive a digital-to-analog converter (DAC) coupled to a speaker to render the audio samples.

Abstract: A computing device may include a multi-functional port in a base housing of the device. The multi-functional port may include an audio input device that can receive and process external audio input signals through an opening in the base housing. The multi-functional port may include a light source that can output light through the opening in the base housing. The multi-functional port may include a microphone boot. The microphone boot may guide external audio input signals into to the audio input device for processing. The microphone boot may also guide light, emitted by the light source, out of the base housing.

Abstract: The disclosure is related to a calibration system for active noise cancellation and a speaker apparatus. The calibration system receives the signals with feedforward control or feedback control active noise cancellation. A gain adjustment element is used to adjust a gain of the signals, and a path selection switch is used to switch connection to a first operational amplifier or to a second operational amplifier. In addition to driving signals, the operational amplifier is also used to adjust a phase of the output signals. The calibration system is able to balance the gain of the signals with active noise cancellation and adjust the phase of signals of a left-channel circuit and a right-channel circuit through gain-phase adjustment. The related speaker apparatus is such as an earphone with the feedforward ANC control circuit, the feedback ANC control circuit, or a hybrid ANC circuit.

Abstract: Acoustic echo cancellation (AEC) processing may be improved by performing echo cancellation using a combined multi-channel reference signal. Two or more reference signals, such as a left and right channel of a stereo source, may be combined and provided to an AEC block configured to receive the combined signal and perform AEC processing using the combined signal. The AEC block may include an adaptive filter that performs operations that cause pre-whitening of the combined reference signal and de-correlation of the individual channels within the combined reference signal. The pre-whitening of the signal flattens the spectrum of the combined reference signal, which may improve convergence speed of the AEC processing in cancelling the echo. The de-correlating of the signal cancels inter-channel correlation between the multiple channels, which may improve convergence speed of the AEC processing in cancelling the echo.

Abstract: In accordance with embodiments of the present disclosure, an information handling system may include an information handling resource, a voltage regulator, a non-transitory computer-readable medium, and a controller. The voltage regulator may be coupled to the information handling resource and configured to deliver electrical energy to the information handling resource. The a non-transitory computer-readable medium may have stored thereon curve fit information, the curve fit information including coefficients of a polynomial for approximating a parameter indicative of the power efficiency of the voltage regulator as a function of a measured output current of the voltage regulator. The controller may be coupled to the voltage regulator and configured to receive information indicative of an output power of the voltage regulator and, based on the information indicative of an output power of the voltage regulator and the curve fit information, calculate an input current of the voltage regulator.

Abstract: The present disclosure provides a method for determining an arrangement of reference sensors for active road noise control (ARNC) in a vehicle with an automatic calibration system. The method includes mounting a plurality of vibrational sensors on a plurality of structure elements of the vehicle to generate a plurality of vibrational input signals and mounting at least one microphone inside a cabin of the vehicle to capture at least one acoustic input signal. The method further includes determining an arrangement of reference sensors from the plurality of vibrational sensors by determining a subset of vibrational sensors which sense the main mechanical inputs of road noise contributing to the at least one acoustic input signal.

Abstract: An acoustic signal processing apparatus includes circuitry to generate, when a plurality of sound receivers receive sound from a plurality of examination directions in a space and outputs acoustic signals of a plurality of channels, an effective signal corresponding to sound coming from each one of the examination directions based on the acoustic signals of the plurality of channels for each one of the examination directions, calculate a feature for each one of the examination directions based on the effective signal generated for each one of the examination directions, and select a target direction from the plurality of examination directions in the space based on the feature calculated for each one of the examination directions.

Abstract: Examples are described relating to an apparatus, a method of manufacture for an apparatus and a system to provide a physical and audible effect. In one exemplary embodiment, the apparatus has an elongate substrate and a plurality of sound producing devices arranged along the elongate substrate.

Abstract: An audio sequencing device includes an input circuit, a prioritizing circuit, a playback circuit, and an output circuit. The input circuit is receives a first audio signal from a first audio source and a second audio signal from a second audio source. The prioritizing circuit assigns a first priority to the first audio signal and a second priority to the second audio signal. The playback circuit sequences at least one of the audio signals based on at least one of an order of receiving the audio signals and the assigned priorities. The output circuit transmits the audio signals to a user device one at a time based on the sequence and the assigned priorities of the audio signals.

Abstract: An electronic controller for an insulin infusion device includes at least one processor device and at least one memory element that cooperate to provide a processor-implemented closed-loop initiation module. The initiation module is operated to obtain a most recent calibration factor for a continuous glucose sensor, the most recent calibration factor representing a first conversion value applicable to convert a first sensor value to a first blood glucose value. The initiation module also obtains a prior calibration factor for the sensor, and calibration timestamp data for the most recent calibration factor and the prior calibration factor. The initiation module regulates entry into a closed-loop operating mode of the insulin infusion device, based on the most recent calibration factor, the prior calibration factor, and the calibration timestamp data.

Abstract: A system has a sound generator (20) that generates superimposed sound to a sound to be manipulated. An error sensor (50) measures sound and outputs a corresponding feedback signal (e?(n)). A signal generator (91) generates a sound signal (y(n)). A controller (92) generates a control signals (?1(n)) and (?2(n)). The adder (94) subtracts one control signal (?2(n)) from the feedback signal (e?(n)) and outputs a modified feedback signal (en(n)) to the signal generator (91). A weighter (95) weights the sound signal (y(n)) with the control signal (?1(n)) and outputs the weighted sound signal (y?(n)). The generated sound signal (y(n)) is a function of the modified feedback signal (e(n)). The controller (92) generates the control signals (?1(n), ?2(n)) such that a value of the amplitudes of the feedback signal (?e?(n)?) corresponds to a predefinable value (?).

Abstract: Aspects of the disclosure relate to computing technologies. In particular, aspects of the disclosure relate to mobile computing device technologies, such as systems, methods, apparatuses, and computer-readable media for improving orientation data. In one embodiment, techniques are described for filtering data associated with a sensor coupled to a computing device, by receiving a signal from the sensor, detecting a change in a variability of a first signal parameter from a plurality of signal parameters from the signal, and adjusting, based at least in part on the detected change in the variability of the first signal parameter, at least one filter parameter of a filter used to filter a second signal parameter from the signal.

Abstract: A removable condenser microphone assembly comprises a large-diaphragm condenser transducer having front and rear diaphragms; a cylindrical harness inside of which the transducer is mounted; a cylindrical ring inside of which the harness is mounted; a connector secured to the bottom of the ring configured to removably secure the microphone assembly to a body of a pencil microphone and to removably electrically couple the microphone assembly to pre-amp circuitry within the pencil microphone body; a switch secured to the ring; and a printed circuit board electrically coupled to the transducer and the switch and, in response to positions of the switch, separately control phantom power to the front and rear diaphragms of the transducer creating a plurality of selectable polar patterns of the transducer.

Abstract: A loudspeaker including a first housing which delimits a acoustic chamber, an acoustic driver disposed within the acoustic chamber, a second housing which delimits a second chamber disposed adjacent to the acoustic chamber, a heat source disposed within the second chamber, a passive radiator disposed in communication with the acoustic chamber and the second chamber, a vent disposed in communication with the second chamber and with an exterior of the loudspeaker, wherein the passive radiator is configured to move in response to a movement of the driver, where the passive radiator is further configured to direct an airflow proximate to the heat source during the movement of the passive radiator and to direct the airflow through the vent to the exterior of the loudspeaker.

Abstract: A system is disclosed for detecting and calculating the level of ambient and/or environmental noise, such as electromagnetic interference generated by electric power lines, ambient lights, light dimmers, television or computer displays, power supplies or transformers, and medical equipment. In some embodiments, the system performs frequency analysis on the interference signal detected by light photodetectors and determines the power of the interference signal concentrated in the analyzed frequency bands. The worst-case interference level can be determined by selecting the maximum from the computed power values. In some embodiments, the determined interference signal power can be compared with the noise tolerance of a patient monitoring system configured to reliably and noninvasively detect physiological parameters of a user. The results of the comparison can be presented to the user audio-visually. In some embodiments, the system can be used to perform spot check measurements of electromagnetic interference.

Abstract: Embodiments of systems and methods are described for estimating a position of a loudspeaker and notifying a listener if an abnormal condition is detected, such as an incorrect loudspeaker orientation or an obstruction in a path between the loudspeaker and a microphone array. For example, a front component of a multi-channel surround sound system may include the microphone array and a position estimation engine. The position estimation engine may estimate the distance between the loudspeaker and the microphone array. In addition, the position estimation engine may estimate an angle of the loudspeaker using a first technique. The position estimation engine may also estimate an angle of the loudspeaker using a second technique. The two angles can be processed to determine whether the abnormal condition exists. If the abnormal condition exists, a listener can be notified and be provided with suggestions for resolving the issue in a graphical user interface.

Abstract: A battery powered, portable multi-channel audio mixing console is disclosed. The mixing console is configured to be easily held in a user's hand like a remote control. The console is configured to wirelessly receive, via separate channels, two or more independent stereo input audio signals from separate audio sources such as cellular phones, iPads, iPods, or other computing or media player devices, independently equalize the frequencies of those input signals, independently adjust the volume of those input signals, and then mix those audio signals into a combined output signal. The console further includes user controls that can vary the relative magnitude of the first and second input signals in the combined output signal. The console is further adapted to transmit the combined output signal to one or more speakers via independent channels.

Abstract: A system includes a sound generator (20) that generates sound superimposed to sound to be manipulated. An error sensor (50) measures superimposed sound and outputs a corresponding feedback signal (e?(n)). A signal generator (91) generates a sound signal (y(n)). A controller (92) generates a control signal (?(n)) representing a value of a sequence of rational numbers. A weighter (93) weights the generated sound signal (y(n)) with the control signal (?(n)) and inverts it. An adder (94) adds the weighted/inverted sound signal to the feedback signal (e?(n)) and outputs a modified feedback signal (e(n)) to the signal generator (91). A weighter (95) weights the generated sound signal (y(n)) with the difference from one and with the control signal (?(n)) and outputs the sound signal y?(n). The generated sound signal (y(n)) is a function of the modified feedback signal (e(n)).