Abstract: A mobile communication terminal including: a window glass provided on a front side of a terminal main body; a speaker hole formed on an edge of the terminal main body corresponding to an edge of an outer portion of the window glass, exposed to the outside, and formed to correspond to a speaker provided in the terminal main body; and a sensor connected to the speaker, and sensing a direction in which the terminal main body is slanted, wherein the sensor controls the speaker to open or close the speaker hole according to the direction in which the terminal main body is slanted.

Abstract: In reverberant environments, reflected waves including an echoic sound and a muffled sound affect and disable recognition of sound arrival directions. As a result, the subjective clearness of the sounds deteriorates. In order to enhance the clearness of a reproduced sound in a reverberant environment, a pre-processing filter unit corrects an input sound signal portion having a frequency band relating to human auditory recognition on a sound wave arrival direction, and speakers reproduce the sound signal. The correction involves attenuating an input sound signal in the frequency band portion, based on the relationship between the frequencies of the input sound signal and the magnitude of influence to the recognition of the sound wave arrival direction. This attenuation is achieved by filtering using filter coefficients that are set by a first filter characteristic setting unit using hearing characteristic parameters that are set by a hearing characteristic setting unit.

Abstract: A method and portable terminal for automatically controlling an output volume of audio output of a head phone when the head phone is connected to a portable terminal are provided. The method of controlling an audio output in a portable terminal, includes outputting a test audio signal to a head phone for a preset playback time when the head phone is connected to the portable terminal; collecting neighboring noise interval information and impedance interval information of the head phone while the test audio signal is output to the head phone; determining impedance of the head phone based on the neighboring noise interval information and the impedance interval information; determining a gain of an audio signal based on the determined impedance; and amplifying the audio signal in proportion to the determined gain, and outputting the amplified audio signal to the head phone.

Abstract: The method comprises the determination of an observation vector that comprises only electrical measurements of the voltage (Umes) at the loudspeaker terminals and of the current (i) passing through the loudspeaker, and a state vector (X) whose components comprise: values of linear parameters of the loudspeaker response such as the electrical (Re) and mechanical (Req) resistance, and polynomial coefficients of nonlinear parameters such as the force factor (Bl), the equivalent stiffness (Keq) and the electrical inductance (Le). The voltage and current measurements are applied to an estimator with a predictive filter of the extended Kalman filter incorporating a representation of a dynamic model of the loudspeaker. This filter operates a prediction of the state vector (X) and readjusts this prediction by calculation of an estimate (Uest) of the voltage based on the state vector and on the measured current and comparison of this estimate with the measurement (Umes) of the voltage.

Abstract: A headset is able to be coupled via a cable to an intercom system, is able to be wirelessly coupled to a wireless device via a wireless transceiver of the headset, and is able to be connected to a wired device via another cable. A controller of the headset separately monitors the microphone conductors and audio conductors by which the headset may be coupled to the intercom system to detect whether or not one or both of a communications microphone and an acoustic driver of the headset are coupled to the intercom system, and monitors the operating state of the wireless transceiver to detect whether or not the wireless transceiver is inactive, on standby or in use; and selectively couples a system ground conductor to one of the microphone conductors, and selectively provides a local microphone bias voltage in response to what is observed through such monitoring.

Abstract: A method of setting a parameter for an audio apparatus including a delay processing unit which applies a delaying process and an amplifying process on an input audio signal in accordance with the set parameter, includes: performing a measuring process including: causing an outputting device to output measurement sound; and measuring a first impulse response at a sound receiving point; performing a specifying process including: analyzing the measured first impulse response; calculating a second impulse response at the sound receiving point when the outputting device outputs sound in case where a parameter is set in the audio apparatus and an audio signal indicative of the measurement sound is input to the audio apparatus; and specifying a parameter in accordance with the calculated second impulse response; and performing a setting process of setting the specified parameter in the delay processing unit.

Abstract: An array speaker system canceling non-uniform radiation patterns and a method for implementing the same are provided. The method for canceling non-uniform radiation patterns includes predicting radiation patterns in an array speaker with respect to input signals, generating cancellation signals with respect to at least one region corresponding to non-uniform radiation patterns of the predicted radiation patterns, synthesizing the input signals and the cancellation signals, and outputting the synthesized signals to the array speaker. Sound signals from which non-uniform radiation patterns having distorted sound are cancelled such that a stable sound field having uniform radiation characteristics may be obtained and provided to a listener.

Abstract: An illustrative embodiment includes a method for use in performing acoustic calibration of at least one audio output device for a plurality of listening locations. An audio input device generates a data signal based on a series of one or more tones output by the at least one audio output device. The audio input device wirelessly transmits the data signal to a calibration device. The audio input device is one of a plurality of audio input devices deployed at respective ones of the plurality of listening locations. The data signal is one of a plurality of data signals generated by respective ones of the plurality of audio input devices based on the series of one or more tones output by the at least one audio output device. The plurality of data signals are wirelessly transmitted by the respective ones of the plurality of audio input devices to the calibration device.

Abstract: An electronic device having an enclosure including an upper panel and a bottom panel operably connected to the upper panel. A transducer is operably connected to the enclosure and the transducer is configured to mechanically vibrate the enclosure. The transducer includes an electromagnet, a magnet in communication with the electromagnet and a bracket substantially surrounding the electromagnet and the magnet, the bracket substantially secures the transducer to the bottom panel.

Abstract: Method, system and computer product are provided for a computer implemented method for providing a confidence measure for speaker diarization. The method includes: receiving an audio session as unsegmented audio data; computing a spectral ratio of principal component analysis (PCA) of sections of the received audio session by a ratio between the largest eigenvalue and the second largest eigenvalue; using the PCA spectral ratio as a confidence measure for speaker diarization processing.

Abstract: The present invention relates to a method and a circuit for testing a tweeter, said tweeter being part of a loudspeaker system, wherein the method includes the steps of: applying a high-frequency voltage signal to one terminal of said tweeter, said high-frequency voltage signal being generated by first electronic means; applying a constant voltage signal to the other terminal of said tweeter, said constant voltage signal being generated by second electronic means; measuring a current Iload that flows through said tweeter into said second electronic means; determining a connect/disconnect state of said tweeter from the value of said current.

Abstract: A signal processing device includes: a sound adjustment amount calculation unit which calculates a sound adjustment amount for adjusting sound characteristics of each channel to a predetermined sound characteristic for each channel, using a sound signal that is obtained by collecting the outputs of each channel; an evaluation value calculation unit which calculates a coefficient allocation evaluation value for allocating a size of a filter coefficient necessary for the sound adjustment of the respective channels for each channel, based on the sound adjustment amount that is calculated by the sound adjustment amount calculation unit; and a filter coefficient calculation unit which calculates the filter coefficient for each channel using the coefficient allocation evaluation value that is calculated by the evaluation value calculation unit.

Abstract: A method consistent with the present invention involves displaying a window on a computer monitor; at one or more programmed processors, determining a position of the window on the computer monitor; at the one or more programmed processors, deducing a user position for a user of the window with the window based on the position of the window on the computer monitor; and steering audio signals from an application running in the window to a loudspeaker in an array of loudspeakers, where the loudspeaker is a loudspeaker in the array of loudspeakers closer to the deduced user position than another loudspeaker in the array. This abstract is not to be considered limiting, since other embodiments may deviate from the features described in this abstract.

Abstract: The application discloses a system and method for playback of audio signals though a large variety of headphone devices. To optimize playback, the specific headphone device is identified so the proper amplifier gain and equalization settings can be applied. Such identification is determined by measuring the headphone impedance as a function of frequency, and comparing the impedance data with data of know devices in a database. Once a match is found, the appropriate audio gain and equalization settings can be applied.

Abstract: The present invention disclosed an all-digital speaker system device based on multi-bit ?-? Modulation, comprising an A/D converter, an interpolation filter, a ?-? modulator, a dynamic mismatch regulator, a differential buffer and a speaker array. The present invention takes advantage of the ?-? modulation technique to effectively reduce the cost and complexity of the hardware realization of the speaker system device, thus realizing the full digitalization of the whole sound transmission link and the integration of the system device with low power dissipation and volume. Furthermore, the system device according to the present invention is provided with a better control ability on the local sound field, thus providing a better practicable method for the voice secret transmission.

Abstract: The invention converts non audio systems into distributed audio sources for active noise control solutions. The system transforms non acoustic structures into soundboards using inertial type acoustic transducers. Acoustic parameters unique for each application due to the variation in properties of the sound board are compensated by equalizers. The invention also uses damping means to limit the reflection of bending waves from the edges. The inertial type acoustic transducer is driven by an amplifier. The acoustic signal to the amplifier is modified by a signal conditioner to compensate for the non optimal response of the acoustic system. An external controller communicates with the amplifier to control its operating parameters. A series of distributed audio sources in a variety of positions may each be addressable as a node on a network wherein noise detected at that source is analyzed and the system generates sound at that source to mask the noise.

Abstract: Devices connected with a computer system are interrogated to detect whether they are configured and functioning. In an audio device implementation, a known audio sample is output to a loudspeaker. If the loudspeaker is properly configured and functional, an attached microphone will capture the sound corresponding to the audio sample generated by the loudspeaker. If sound corresponding to the audio sample is detected, an indication is provided to the user that the microphone and loudspeaker are operational. If no sound corresponding to the audio sample is detected, an indication is provided to the user that the loudspeaker is not enabled and additional configuration is required.

Abstract: According to one embodiment, a measuring apparatus includes: an output module configured to temporally exclusively output, to a measuring target system, a first output signal corresponding to a first measuring signal for sweeping frequency and a second output signal corresponding to a second measuring signal for sweeping frequency and having a different amplitude characteristic from an amplitude characteristic of the first measuring signal; and a frequency characteristic computation module configured to synthesize a first frequency amplitude spectrum obtained from a first reception signal when a sound output from the measuring target system based on the first output signal is received and a second frequency amplitude spectrum obtained from a second reception signal when a sound output from the measuring target system based on the second output signal is received to generate frequency characteristic data representing an acoustic characteristic.

Abstract: A thermal control module computes an estimate of a temperature of a speaker, based on an audio signal that is driving the speaker, and computes a gain that is applied to attenuate the audio signal to prevent overheating of the speaker. Thermal control module computes an adapted impedance, being an estimate of the speaker's impedance including its DC resistance, and uses it to compute the temperature estimate. The adapted impedance is obtained from a normal adaptation process when a measured voltage of the speaker is above a threshold, and a decay process when the measured voltage is below the threshold. Other embodiments are also described.

Abstract: An electronic audio system that determines matching and non-matching speakers is disclosed. The electronic audio system includes a first speaker; a second speaker; and a comparator circuit. The comparator circuit is coupled to the first speaker and the second speaker to receive an input signal from both the first and second speaker. The comparator circuit is configured to determine if the first speaker and the second speaker are matching speakers or non-matching speakers, in which, the first and second speakers are matching speakers if they are common vendor speakers. Other embodiments are also described and claimed.

Abstract: A power saving system includes a sensor, a processor, a timer, and a switch. The sensor senses movements in the diaphragm of the loudspeaker. The processor determines whether the loudspeaker is power-on and in use or power-on but not in use. The processor further determines whether a preset period of time has elapsed since the indication of power-on but not in use was first given. If the preset period of time has elapsed, the processor further controls the switch to disconnect the power source from the loudspeaker.

Abstract: An example set-up method for a loudspeaker system capable of generating at least one directed beam of audio sound includes emitting directional beams of set-up sound signals from the loudspeaker system into a room, registering at least one reflection of the emitted signals at one or more locations within the room, and evaluating the registered reflected signals to obtain data for use in configuring the surround sound system.

Abstract: An electrical apparatus includes a frame, a speaker connected to the frame, a digital signal processor in communication with the speaker to receive audio data and control data to control the speaker, the digital signal processor connected to the frame, and a lamp base coupler electrically connected to the speaker and receiver, the lamp base coupler detachably connectable to a power source, when the power source is present. A method of steering the diffused sound field includes, broadcasting at least one calibration audio signal through a plurality of speakers (M) in an audio system, receiving the calibration audio signal in a plurality of microphones spaced apart and positioned about at a listening position, and calculating respective relative speaker placement angles relative to the listening position between each of the plurality of speakers in response to receipt of the calibration audio signal in the plurality of microphones.

Abstract: A loudspeaker drive circuit comprises a signal path compressor/limiter (12) for implementing a change to the peak-mean amplitude ration in the time domain. A feedforward control loop measures an acoustic quality of the signal at the output of a control loop compressor/limiter (20) or the output of the signal path compressor/limiter (12), and also estimates a loudspeaker excursion based on the signal at the output of the control loop compressor/limiter (20). The signal path compressor/limiter is controlled based on the acoustic quality measurement and excursion estimation. The invention provides a method for the maximisation of the acoustic output of a loudspeaker by adjusting the characteristics of a compressor/limiter, with the constraint that the audio quality stays acceptable, and that the diaphragm displacement does not exceed a certain threshold.

Abstract: Systems and methods for calibrating a loudspeaker with a connection to a microphone located at a listening area in a room. The loudspeaker includes self-calibration functions to adjust speaker characteristics according to effects generated by operating the loudspeaker in the room. In one example, the microphone picks up a test signal generated by the loudspeaker and the loudspeaker uses the test signal to determine the loudspeaker frequency response. The frequency response is analyzed below a selected low frequency value for a room mode. The loudspeaker generates parameters for a digital filter to compensate for the room modes. In another example, the loudspeaker may be networked with other speakers to perform calibration functions on all of the loudspeakers in the network.

Abstract: A harmonic echo power estimator estimates power of echo generated by harmonic loudspeaker nonlinearities in a user equipment having an echo path between a loudspeaker input and a microphone output. The estimator includes a frequency band mapper that maps each frequency band in a set of loudspeaker output signal frequency bands into a corresponding array of loudspeaker input signal frequency bands, where each frequency band in the set is mapped into several frequency bands in the corresponding array. A power estimator determines a power estimate of each input signal in each array of frequency bands. A power estimate combiner combines determined power estimates in each array of frequency bands into a corresponding estimate of loudspeaker input power generating harmonic loudspeaker nonlinearities. A power estimate transformer transforms the estimates of loudspeaker input power across the echo path into power estimates of the echo generated by the harmonic loudspeaker nonlinearities.

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: The present invention relates to a method and a circuit for testing a tweeter, said tweeter being part of a loudspeaker system, wherein the method includes the steps of: applying a high-frequency voltage signal to one terminal of said tweeter, said high-frequency voltage signal being generated by first electronic means; applying a constant voltage signal to the other terminal of said tweeter, said constant voltage signal being generated by second electronic means; measuring a current Iload that flows through said tweeter into said second electronic means; determining a connect/disconnect state of said tweeter from the value of said current.

Abstract: A system for sensing an operational status or level of performance of an acoustic horn makes use of a vibration sensor that is operatively coupled to the driver of the acoustic horn. The sensor can be an accelerometer. The sensor outputs a vibration signal when the acoustic horn is operating. Characteristics of the vibration signal are compared to threshold values or to patterns to determine the operational status or level of performance of the acoustic horn.

Abstract: A method of producing a plurality of broadband response files derived from empirical testing on at least one human subject, for convolving with an audio input signal represented as digital samples to produce a stereo output signal (having a left field and a right field) such that the stereo signal emulates the production of said audio signal from a specified audio source location relative to a listening source location. A reference signal is derived for each of a plurality of test positions. An originating signal is derived from at least one reference signal. Each reference signal is deconvolved with the originating signal to produce a broadband response file for each test position. A data storage facility having a plurality of broadband response files stored therein.

Abstract: A method and apparatus for reducing resonance of a loudspeaker by using a model derived from a software simulated loudspeaker are provided. The method includes generating a model to determine resonance characteristics of the loudspeaker based on physical parameters of the audio reproducing device; determining accuracy of the model of the loudspeaker by comparing calculated frequency characteristics of the model of the loudspeaker and measured frequency characteristics of the loudspeaker that is outputting sound; generating a resonance reduction filter based on the model of the loudspeaker according to the determined accuracy; and reducing the resonance characteristics of the loudspeaker by applying the resonance reduction filter to input audio data.

Abstract: An in-ear audio device has a casing on which is disposed one or more bags are positioned to be filled with fillable material during a customization process in which a test sound is acoustically output into an ear canal by an acoustic driver of the in-ear audio device while the one or more bags are being filled, and a microphone acoustically coupled to the ear canal is employed to detect sounds within the ear canal that are indicative of the frequency response of the acoustic output of the acoustic driver to determine when the degree of sealing of the ear canal by the one or more bags is sufficient to achieve a desired quality of frequency response.

Abstract: A method, hearing aid, and computer product for changing the sequence of program positions of hearing programs are provided. The program positions of the hearing programs have a predefinable sequence. Time periods, within which hearing programs are selected, are measured. A new sequence of the program positions is calculated taking the measured time periods into account. The sequence of the program positions after a predefinable time span is modified in accordance with the calculated sequence. As a result hearing aids adapt to changed environments and place favorite programs in the front section of the program positions in accordance with the hearing aid wearer's preference.

Abstract: An apparatus and method for interfacing with an earphone are provided. The method includes checking if the earphone is uninstalled promptly after the connection of the earphone is sensed in an external port and if it is sensed that the earphone is uninstalled, continuing to apply a voltage to a MIC_BIAS terminal. The generation of a noise of the MIC_BIAS terminal caused by applying of a periodic voltage resulted from loose insertion of the earphone is prevented.

Abstract: A loudspeaker amplifier drive circuit performs analysis of various signals relating to the input and output of the amplifier such that the characteristics and/or diagnostics of a loudspeaker driven by the amplifier can be derived. These are then presented as outputs, so that different circuitry can make use of the information for audio signal processing.

Abstract: The application relates to a test device for a speaker module of a listening device, the speaker module comprising a speaker unit for converting an electric output signal to an output sound. The application further relates to a test system and to a data update device. An object of the present application is to provide an alternative scheme for identifying a receiver (speaker) in a listening device, e.g. a hearing aid. The problem is solved in that the test device being a separate device adapted for being electrically connected to the speaker module in a test situation, but not during normal operation of the listening device. An advantage of the invention is that it provides a flexible alternative to a receiver identification solution that is integrated into the listening device.

Abstract: A signal-processing circuit for the generation of a loudspeaker signal may include an input for the feeding of a digital input signal and a digital equalizer filter that is coupled with the input and has at least one first recursive filter that is defined by a first adjustable set of coefficients, an amplification device, a current measurement device, a digital filter block that may be configured to generate, through filtering, an estimate signal as a function of the measurement signal, and a matching block that may be configured to determine a new set of coefficients as a function of the measurement signal, the estimate signal, and the filtered input signal.

Abstract: A system for compensating and driving a loudspeaker includes an open loop loudspeaker controller that receives and processes an audio input signal and provides an audio output signal. A dynamic model of the loudspeaker receives the audio output signal, and models the behavior of the loudspeaker and provides predictive loudspeaker behavior data indicative thereof. The open loop loudspeaker controller receives the predictive loudspeaker behavior data and the audio input signal, and provides the audio output signal as a function of the audio input signal and the predictive loudspeaker behavior data.

Abstract: A load detection technique for a load comprising multiple frequency-dependant sub-loads comprises measuring a representation of the impedance characteristic of the load; providing stored representations of a multiplicity of impedance characteristics of the load; each one of the stored representations represents the impedance of the load when at least a particular one of the sub-loads is in a fault condition; and comparing the measured representation of the current impedance characteristic of the load with each one of the stored representations and in case that the measured representation matches a stored representation, identifying the sub-load or sub-loads being in a fault condition by the corresponding stored representation.

Abstract: A sound quality testing device for testing a communication apparatus has a sound generating unit and a sound receiving unit. The sound quality testing device includes a carrying unit, a first testing module, and a second testing module. The carrying unit carries the communication apparatus. The first testing module generates and sends a sound signal to the sound receiving unit. The second testing module receives a sound-generating signal generated by the sound generating unit. The sound quality testing device provides a standardized simulation testing environment having low or no noise signals, such that the communication apparatus can be tested precisely and steadily in terms of sound quality, such as volume, frequency responses, and harmonic wave distortion.

Abstract: A system for headphone equalization includes a stored set of predetermined tone burst reference signals and a stored set of predetermined tone burst test signals that form a range of frequencies used in a user specific audio test to develop a headphone correction filter. A predetermined tone burst reference signal and a predetermined tone burst test signal may intermittently and sequentially drive a transducer included in the headphone. A loudness of the predetermined tone burst reference signal may be fixed and a loudness of the predetermined tone burst test signal may be variable with a gain setting. The gain setting may be used to generate the headphone correction filter.

Abstract: To efficiently detect a standing wave generated in a room, a standing-wave detection apparatus for detecting a standing wave in a predetermined space, comprises: a sound-receiving unit adapted to receive a sound generated from a sound source arranged in the predetermined space; a storage unit adapted to store time series sound pressure level data acquired by the sound-receiving unit during movement along a path in the predetermined space; an adjustment unit adapted to adjust the time series sound pressure level data stored in the storage unit, based on an adjustment curve determined using a lower envelope of the time series sound pressure level data stored in the storage unit; and a detection unit adapted to detect an existence position of a standing wave in the predetermined space based on the adjusted time series sound pressure level data.

Abstract: A piezoelectric speaker device that includes an insulation layer having an electric insulation property formed on a user-side driving electrode, and flaw detection electrode lines formed on the insulation layer. If the flaw detection electrode lines are determined to be damaged due to the occurrence of a flaw, the driving voltage applied to the driving electrodes is lowered. In order to prevent electric-shock accidents, the driving voltage is lowered to below 42.4 V, preferably, and to 0 V, more preferably.

Abstract: A sound system for capturing and reproducing sounds produced by a plurality of sound sources. The system comprises a device for receiving sounds produced by the plurality of sound sources and converting the separately received sounds to a plurality of separate audio signals without mixing the audio signals. The system may further comprise a device for separately storing the plurality of separate audio signals on a recording medium without mixing the audio signals and a device for reading the stored audio signals from the recording medium. The system further includes a reproduction system for recreating the plurality of separate audio signals. Also, the system comprises an amplification network which comprises a plurality of amplifier systems, with one or more separate amplifiers in each amplifier system for separately amplifying each of the separate audio signals.

Abstract: Optimized sound waves presented to the listener by headphones, notwithstanding differences in ear geometry and headphone positioning. A test signal causes an acoustic sensor to receive sound waves actually formed in the listener's ear cavity. A response from the sensor is compared with an expected ear cavity transfer function, from which desired adjustments to the audio signal are determined. The audio signal might be received from an application program, calibrated by an interface software element, and adjusted thereby, before forwarding to the headphones. Calibration might be performed from when the headphones are positioned, or dynamically in response to changes in the transfer function.

Abstract: An acoustic apparatus includes: a plurality of speakers, which are connected in parallel to each other; and a line break detector for detecting the number of line break in the speakers. The line break detector includes: a detection voltage generator for applying a reference voltage to the speakers; and a line break determination device for determining the number of line break according to an actual voltage of the speakers when the reference voltage is applied to the speakers or when application of the reference voltage to the speakers is stopped. Thus, without using an impedance measurement device, the number of line break is calculated based on the actual voltage of the speakers. The number of line break in the speakers is easily detected with low manufacturing cost.

Abstract: An ambient noise removal device includes: a sound signal output section that outputs a sound signal to thereby drive a speaker; a noise detection section that detects an electromotive force produced in the speaker by ambient noise; and a noise cancellation section that generates, on the basis of the electromotive force, a signal of an opposite phase from the ambient noise and adds the generated signal to the sound signal to thereby cancel the ambient noise.

Abstract: This invention concerns a loudspeaker driver includes at least one actuator connected to a vibrating support to impart excitation to the latter when caused to move, wherein the loudspeaker driver further includes a plurality of sensing members arranged to move with the at least one actuator, each sensing member providing output sensing data dependent on the velocity of said at least one actuator, and means for determining the position of the at least one actuator based on at least one ratio (X/Y) of output sensing data or of linear combinations of output sensing data provided from the plurality of sensing members, said at least one ratio being independent of the velocity of the at least one actuator.

Abstract: An audio signal processing apparatus includes: a test signal supply unit to supply a test signal to each speaker of a multi-channel speaker including a center speaker and others; a speaker angle calculation unit to calculate an installation angle of each speaker with an orientation of a microphone as a reference, based on test audio output from each speaker and collected by the microphone; a speaker angle determination unit to determine an installation angle of each speaker with a direction of the center speaker from the microphone as a reference, based on the installation angle of the center speaker and the installation angles of the other speakers with the orientation of the microphone as a reference; and a signal processing unit to perform correction processing on an audio signal based on the installation angles of the speakers with the direction of the center speaker from the microphone as a reference.

Abstract: An electrical apparatus to sense current through a load includes a first input terminal having a first input voltage relative to a reference, a second input terminal having a second input voltage relative to the reference, a first load terminal of the load having a first load voltage relative to the reference, a second load terminal of the load having a second load voltage relative to the reference, a first current sensing element connected between the first input terminal and the first load terminal and a second current sensing element connected between the second input terminal and the second load terminal. A first sense voltage is determined by a relationship between the first input voltage and the second load voltage and a second sense voltage is determined by a relationship between the second input voltage and the first load voltage.