Abstract: An integrated circuit for processing audio and voice signals in a telephone is disclosed. The integrated circuit comprises an audio output port having a first and a second output terminal for connection to a first and a second input terminal, respectively, of a speaker. The integrated circuit further comprises an audio input port having a first and a second input terminal for connection to a first and a second output terminal of an audio input source. The integrated circuit also comprises at least one audio output path and one audio input path. Moreover, the integrated circuit comprises a test impedance, such as a resistor, (RT) for measuring an electrical impedance of the speaker.

Abstract: A audio output device includes: a speaker position detecting unit which detects the position of a speaker; a masking sound producing section which produces a masking sound; a plurality of loudspeakers which output the masking sound; and a localization controlling section which controls a localization position of the masking sound based on the speaker position detected by the speaker position detecting unit, and which supplies a sound signal relating to the masking sound to at least one of the plurality of loudspeakers.

Abstract: An improved media delivery apparatus is disclosed. The media delivery apparatus can, for example, be a portable speaker system. In one embodiment, the media delivery apparatus is used with a portable media player. According to one aspect, audio output characteristics of a media delivery apparatus can be controlled based on whether a removable component is attached or detached. Another aspect pertains to improved techniques to attach a removable component to a media delivery apparatus. According to still another aspect a portion of a housing of an electrical device, such as a media delivery apparatus, can be utilized as part of a connector or plug.

Abstract: A howling suppression device that can reduce quality deterioration of processed sound includes: a delay unit delaying the input signal to output the delayed input signal as a reference signal; a signal separation unit including an adaptive filter extracting a periodic signal component from the reference signal by adaptively updating a filter coefficient; a howling detection unit detecting an occurrence of howling using at least a signal of the periodic signal component output from the adaptive filter; and a howling suppression unit. The howling suppression unit includes: a suppression filter obtaining the updated filter coefficient from the adaptive filter with timing when the howling detection unit detects the occurrence of the howling, to extract the periodic signal component from the reference signal based on the filter coefficient; and a subtractor subtracting the periodic signal component from the input signal so as to output a signal obtained by the subtraction.

Abstract: A channel divider that can suitably set a crossover frequency when a biwiring-connectable multiway speaker system including network circuits are used is provided. The channel divider includes a low-pass filter LPF and a high-pass filter HPF for dividing a band of a sound signal into a first output signal and a second output signal so as to output the signals, and a control circuit for controlling the filters. The channel divider has a sound output mode in which the control circuit sets a cutoff frequency fcL of LPF to a value higher than fc0 by about ? to 1 Oct., sets a cutoff frequency fcH of HPF to a value lower than fc0 by about ? to 1 Oct., and outputs the first output signal and the second output signal including frequency bands fcL to fcH when any crossover frequency fc0 for defining the band division is specified.

Abstract: One or more acoustic transducers of a device may be adjusted based on automatic detection of device proximity to the user. In a mobile telephone, when the user is using the receiver and holding the telephone against his/her ear, if the telephone detects that the user has moved the telephone further from his/her ear, the telephone will raise the receiver volume. Similarly, if the user is using the speaker, the telephone will adjust the speaker volume as user distance from the telephone changes. In another embodiment the telephone may fade between the receiver and the speaker. Volume is not the only acoustic property that could be adjusted according to user proximity. Frequency response is another property that could be adjusted, such as using appropriate electronic filtering, or by turning on another transducer that is not otherwise being used.

Abstract: A power bank apparatus with speaker combines the function of power bank and the function of speaker. The power bank apparatus not only charges the portable electronic apparatuses but also supplies power to the internal speaker. The voice or music of the portable electronic apparatus is amplified by the speaker of the power bank apparatus to improve the quality of the voice or music.

Abstract: It is possible to provide a frequency characteristics measuring device which can simplify the configuration for performing a measurement and reduce the undue effort required for the measurement. A spectrum analyzer (10) includes: two sets of measuring units having mixers (110, 210), local oscillators (112, 212), and IF sections (120, 220) for separately measuring frequency characteristics of two input signals; a trigger generation section (310) which generates a trigger signal for specifying a measurement start timing in each of the two sets of measuring units; a sweep control section (300) which simultaneously sends an instruction to the two local oscillators (112, 212) when a trigger signal is inputted and performs a sweep control so that the two local oscillators (112, 212) output local oscillation signals of the same frequency at the same timing.

Abstract: An apparatus for changing an audio scene has a direction determiner and an audio scene processing apparatus. The audio scene has at least one audio object having an audio signal and associated meta data. The direction determiner determines a direction of a position of the audio object with respect to a reference point based on the meta data of the audio object. Further, the audio scene processing device processes the audio signal, a processed audio signal derived from the audio signal or the meta data of the audio object based on a determined directional function and the determined direction of the position of the audio object.

Abstract: A method and a device for sound field reproduction from a first audio input signal uses a plurality of loudspeakers for synthesizing a sound field within a preferred listening area in which none of the loudspeakers are located and therefore described as emanating from a virtual source. The method further includes the steps of calculating a plurality of positioning filter coefficients using virtual source description data and loudspeaker description data according to a sound field reproduction technique; and modifying the first audio input signal using the positioning filter coefficients to form second audio input signals. A loudspeaker ranking of the importance of each loudspeaker for the synthesis of the sound field within the preferred listening area is therefore defined. Then, second audio input signals are modified according to the loudspeaker ranking to form third audio input signals. Finally, the loudspeakers are alimented with the third audio input signals which synthesize a sound field.

Abstract: Disclosed are an apparatus and a method for tracking locations of a plurality of sound sources. According to the apparatus and the method, a task for searching sound source candidates is repeated at respective predetermined frames of microphone signals to collect sound source candidates, and only the collected sound source candidates are verified through beamforming, thereby more rapidly and accurately tracking the plurality of sound sources in spite of using a small number of microphones.

Abstract: To provide a speaker array apparatus and a method for setting audio beams in a speaker array apparatus, in which the degree of freedom in the place where the speaker array apparatus is installed is high, and a user can set audio beams easily. A speaker array apparatus 1 sweeps a range of from 0 degree to 180 degrees in front of a speaker array 10 with audio beams based on an audio signal limited to a band where the angles of the audio beams can be adjusted. The speaker array apparatus 1 collects direct sounds or reflected sounds of the audio beams through a nondirectional microphone 2. The speaker array apparatus 1 analyzes the collected audio data, detects peaks not lower than a threshold value, and checks symmetry among the peaks. When there is a symmetry, the angles where the peaks were detected are set as angles with which audio beams of respective channels of a surround-sound should be output.

Abstract: A speaker module of a portable terminal and a method of execution of a speakerphone mode using the same are provided. The speaker module includes a nondirectional microphone, a directional microphone, at least one speaker, and a controller. The nondirectional microphone is applied to one side of the portable terminal. The directional microphone is applied to another side of the portable terminal. The at least one speaker is installed in the same direction near the directional microphone. When the portable terminal is in a speakerphone mode, the controller controls to detect voice signal levels and/or frequency characteristics input to the two microphones, sets a transmission/reception mode of the portable terminal, and mute-processes a corresponding microphone.

Abstract: A failure detection device for a vehicle speaker includes a signal generator, an amplifier, a coupling capacitor, a detection circuit, and a determination section. The signal generator generates a sound signal corresponding to a sound outputted from the speaker. The amplifier amplifies the sound signal generated by the signal generator. The coupling capacitor supplies the sound signal amplified by the amplifier to the speaker. The detection circuit directly or indirectly detects a terminal voltage on a terminal of the speaker. The determination section determines whether an open-circuit occurs in the speaker based on a phase difference between a phase of the sound signal and a phase of the terminal voltage. The terminal of the speaker is coupled to the coupling capacitor.

Abstract: Electronic devices and accessories for electronic devices such as headsets are provided. The electronic devices may produce audio output. The headsets may include earbuds with speakers that play the audio output for a user while the earbuds are located in the user's ears. Circuitry in an electronic device and a headset may be used in evaluating how well the earbuds are sealed to the user's ears. In response to seal quality measurements, informative messages can be generated for the user, overall earbud volume may be increased, balance adjustments may be made to correct for mismatched balance between left and right earbuds, equalization settings may be adjusted, and noise cancellation circuitry settings can be changed. Electrical impedance measurements and acoustic measurements can be used in evaluating seal quality.

Abstract: A sound processing apparatus according to the present invention acquires a test signal for measuring a standing wave state emitted in a listening room, and determines a peak position or a dip position due to a standing wave based on frequency characteristics of the test signal. Next, the sound processing apparatus emits a burst signal corresponding to the frequency of the peak position or the dip position, and acquires this signal. The sound processing apparatus calculates an increment ?P of the acquired signal, which indicates an amount of increase of a peak in the trailing edge portion corresponding to the end position of the burst signal relative to a peak in the portion corresponding to the stationary portion of the burst signal, and attenuates the frequency of the above peak position or dip position of a sound signal to be output by an attenuation depending on ?P.

Abstract: A speaker with a digital signal processor is disclosed. In one aspect, a speaker comprises at least one electromechanical transducer configured to convert an electrical audio signal into sound and a digital signal processor configured to process an audio signal and send the processed audio signal to the electromechanical transducer directly or indirectly.

Abstract: A method of determining the harmonic and anharmonic portions of a response signal (RS) of a device (2), e.g. an electro-acoustic or electric device, comprises the steps of supplying an input signal (IS) to the device (2) causing the device (2) to respond with a response signal (RS), wherein the input signal (IS) is a sinusoidal signal having a continuously increasing or decreasing frequency (f), capturing the response signal (RS), transforming the captured response signal (RS) from the time space into the phase space and analyzing the phase space transformed response signal (TRS) in respect of its harmonic and/or anharmonic portions or establishing reference functions when defining the input signal (IS) and analyzing the response signal (RS) by Fourier transformations carried out by numerical integrations of the reference functions.

Abstract: Presented herein is a system for transmitting sounds to an in-utero infant, that is worn by a pregnant woman, said system comprising a flexible carrier with two arms and adapted to accommodate the continuously changing shape of the abdomen of a pregnant woman; an array of speakers disposed on the arms of the carrier and substantially surrounding the in-utero infant with the sounds emitted from said speakers; an amplifier operatively associated with the array of speakers that enables tailoring the frequencies and tonal qualities of the sounds emitted from the array of speakers to levels suitable for an in-utero infant; and an audio device, operatively connected to the array of speakers and the amplifier, that provides a source from which sounds are played back.

Abstract: An apparatus for measuring a plurality of loudspeakers arranged at different positions includes a generator of a test signal for a loudspeaker; a microphone device configured for receiving a plurality of different sound signals in response to one or more loudspeaker signals emitted by one of the loudspeakers in response to the test signal; a controller for controlling emissions of the loudspeaker signals by the loudspeakers and for handling the different sound signals so that a set of sound signals recorded by the microphone device is associated with each loudspeaker in response to the test signal; and an evaluator for evaluating the set of sound signals for each loudspeaker to determine at least one loudspeaker characteristic for each loudspeaker and for indicating a loudspeaker state using the at least one loudspeaker characteristic. This scheme allows automatic, efficient and accurate measurement of loudspeakers arranged in a three-dimensional configuration.

Abstract: Systems and methods for measuring noise exposure associated with use of a wireless headset are presented. In one example, a transition from a wireless headset standby mode operation to a wireless headset active mode operation is identified. A stored noise dose measurement at the wireless headset is recalled, and a current noise dose measurement is calculated at the wireless headset for a duration of the active mode operation. A transition from the wireless headset active mode operation to the wireless headset standby mode operation is identified, and an updated noise dose measurement is recorded at the wireless headset.

Abstract: In an acoustic design support apparatus, a speaker selection supporter selects a desired speaker as a candidate for use in a given space based on shape information representing a shape of the space. A speaker mounting angle optimizer calculates an optimal mounting direction of the selected speaker by selecting a mounting direction pattern which minimizes a degree of variation among sound pressure levels at a plurality of positions on a sound receiving surface defined in the space. An acoustic parameter calculator calculates a variety of acoustic parameters at sound receiving points within the space based on both of the shape information of the space and the optimal mounting direction of the speaker.

Abstract: There is provided a calibration method for calibrating a variable number of speakers. The method includes determining physical features around a location of each of the variable number of speakers and calibrating at least one of the variable number of speakers.

Abstract: A thermally regulated amplifier system includes an amplifier unit, a temperature-sensing unit and a controller. The amplifier unit includes a power amplifier that has an adjustable gain function. The controller receives temperature readings from the temperature-sensing unit, computes the gain G(n) of the amplifier unit, and provides the computed gain of the amplifier G(n) to the power amplifier unit.

Abstract: The invention relates to an audio device comprising: an audio source arranged to produce audio signals; an audio output arranged to connect at least one speaker with said audio device; impedance calculation means arranged to calculate an impedance of said at least one speaker; a speaker identification module arranged to determine the identity of said at least one speaker using said impedance and identification data relating to one or more reference speakers. Once the identity of a speaker is known, it can be used for audio compensation or marketing information.

Abstract: An audio apparatus is provided. The audio apparatus includes at most one electroacoustic transducer; and an audio controller, coupled to the electroacoustic transducer, for actively controlling the electroacoustic transducer to function as a loudspeaker or a microphone, wherein the loudspeaker converts output electrical signals to output sounds, and the microphone converts input sounds to input electrical signals.

Abstract: An active noise reduction system is provided for receiving an audio input signal and a noise interference signal and calculating an audio broadcasting signal according to a Feedback Filtered-X Least-Mean-Square (FFXLMS) algorithm, wherein the FFXLMS algorithm optimizes a (convergence factor) ? so as to decrease the numbers of divisions operated by the active noise reduction system and increase the operation speed of the active noise reduction system.

Abstract: The method for checking loudspeakers comprises the steps of emitting a pulse over an electrical connection with the loudspeaker; measuring the electrical resistance over the connection; determining the state of the loudspeaker according to the measured resistance; and triggering a corrective action depending on the determined state of the loudspeaker. The claimed method further comprises the steps of associating a plurality of amplification systems to different subsets of the plurality of loudspeakers; assigning different signals to be emitted by different loudspeakers for each of a plurality of control signals; and storing parameters representing the signals to be emitted in each amplification system. When a control signal common to the different amplification systems is received, the amplification systems cause the loudspeakers to emit the different signals and the corrective action comprises memorizing the parameters representing the status of the loudspeaker.

Abstract: According to one embodiment, an information processing apparatus includes a measurement module and a correction filter design module. The measurement module measures frequency characteristics of an earphone connected to an output terminal using a measurement audio signal output from the output terminal and collected by a microphone. The correction filter design module designs a correction filter in association with one range of a treble range higher than a crossover frequency range and a bass range lower than the crossover frequency range, based on the measured frequency characteristics of the earphone and predetermined goal frequency characteristics.

Abstract: A method of measuring an impulse response of an amplifier coupled in operation to a loudspeaker arrangement includes: (a) coupling directly to a connection between the amplifier and the loudspeaker arrangement for obtaining access to a drive signal (Samp) applied to the loudspeaker arrangement to generate an acoustic output (S2); (b) disposing a microphone arrangement for receiving the acoustic output (S2) of the loudspeaker arrangement; (c) using a test signal generator to apply a test signal (Ssw) to an input of the amplifier; and (d) receiving at a digital signal processing arrangement (DSP, 210) at least the drive signal (Samp) and the acoustic output (S2) corresponding to the test signal (Ssw) and performing on these signals a signal processing operation for determining an impulse response for at least one of: the amplifier, the loudspeaker arrangement.

Abstract: A portable handheld test fixture has an acoustic port, e.g. a speaker port, a microphone, and/or an earpiece/receiver port. Once the media device has been installed onto the test fixture, the port becomes acoustically coupled to a respective acoustic aperture of the media device that is associated with a speaker, a microphone, or an earpiece/receiver as the case may be. A sound test tool, e.g. a reference sound source and/or a sound pressure level, SPL, meter, has an input or output sound port that can be removably coupled to the acoustic port of the test fixture. Other embodiments are also described and claimed.

Abstract: An apparatus includes a capacitor, a current generating circuit communicatively coupled to the capacitor, and a current pulse timing circuit communicatively coupled to the current source circuit. The current timing pulse circuit is configured to time durations of a first plurality of current pulses from the current generating circuit for charging the capacitor and a second plurality of current pulses for discharging the capacitor, and step the durations of the current pulses between a minimum duty cycle and a maximum duty cycle. A cycle of providing the first plurality of current pulses and providing the second plurality of current pulses results in generation of a subsonic pseudo-sinusoidal pulse signal at the capacitor.

Abstract: A frequency equalization system is provided that substantially equalizes the room frequency responses generated by at least one loudspeaker within a listening area so that the frequency responses in the listening area are substantially constant and flat within a desired frequency range with minimum signal latency. The frequency equalization system may use multiple microphones to measure the audio signals of one or more subwoofers to achieve an improved bass response that is flat across the relevant frequency range. The system employs an algorithm that is a closed-form, non-iterative, mathematical solution and features short computation time with reduced delays.

Abstract: The invention provides an electronic apparatus. In one embodiment, the electronic apparatus comprises a hard disk, a speaker, and a processor. The hard disk comprises a vibration protection mechanism which lowers the performance of the hard disk when the vibration protection mechanism is activated. The speaker broadcasts a sound. The processor determines whether a vibration level of the speaker is greater than a threshold level, determines whether data stored in the hard disk is being accessed when the vibration level of the speaker is greater than the threshold level, and lowers a volume of the speaker when the data stored in the hard disk is being accessed and the vibration level of the speaker is greater than the threshold level, thereby preventing the vibration protection mechanism from being activated to maintain the performance of the hard disk.

Abstract: A method and apparatus for detecting a malfunctioning speaker are provided. The apparatus detects a speaker abnormality by identifying the state of the speaker when turning the power on or off, switching from standby mode to play mode, or performing testing. The apparatus generates an audio signal having a frequency higher than a particular frequency, outputs the audio signal to the speaker, and determines that the speaker is malfunctioning if the level of a fed-back frequency signal is lower than a reference level. If the speaker is determined to be malfunctioning, the power supply is cut and the danger of a fire starting is accordingly reduced.

Abstract: Embodiments of the invention include methods, apparatus, and systems for detecting a predicted future or current failure of a microphone of a headset. The failure may have been caused by organic matter buildup creating a signal path or short circuit across the microphone's circuitry. The headset is connected to a mobile device having a network interface that is used to send a notification message to a remote supply management system server. A failure detection circuit detects the failure based on a decrease in a microphone bias signal or increase in headset temperature over time. In some cases, the failure is based on an increase in a microphone bias signal over time. Upon detection of the failure, it signals that a failure notification be transmitted to the remote supply management system. The notification may then cause a new headset to be sent to the owner of the mobile device. Other embodiments are also described and claimed.

Abstract: A loudspeaker drive circuit uses a dynamic range compressor to implement a non-linear gain function between the input signal to the dynamic range compressor and an output signal from the dynamic range compressor. The output is used to drive a loudspeaker, and the operating parameters of the dynamic range compressor are varied in dependence on a criterion, such as the estimated voice coil temperature, the power consumption or the acoustical distortion.

Abstract: At least one exemplary embodiment is directed to a new spatial audio enhancing system including a novel Adaptive Sound Upmixing System (ASUS). In some specific embodiments the ASUS provided converts a two-channel recording into an audio signal including four channels that can be played over four different loudspeakers. In other specific embodiments the ASUS provided converts a two-channel recording into an audio signal including five channels that can be played over five different loudspeakersn even other specific embodiments the ASUS provided converts a five-channel recording (such as those for DVD's) into an audio signal including eight channels that can be played over eight different loudspeakers. More generally, in view of this disclosure those skilled in the art will be able to adapt the ASUS to process and provide an arbitrary number of audio channels both at the input and the output.

Abstract: In an acoustic design support apparatus, a speaker selection supporter selects a desired speaker as a candidate for use in a given space based on shape information representing a shape of the space. A speaker mounting angle optimizer calculates an optimal mounting direction of the selected speaker by selecting a mounting direction pattern which minimizes a degree of variation among sound pressure levels at a plurality of positions on a sound receiving surface defined in the space. An acoustic parameter calculator calculates a variety of acoustic parameters at sound receiving points within the space based on both of the shape information of the space and the optimal mounting direction of the speaker.

Abstract: A detecting circuit measures at least one response characteristic of an output channel in an electro-acoustic transducer system. A memory stores a plurality of equalizations, each equalization corresponding to a known electro-acoustic transducer system associated with at least one response characteristic stored in the memory. A processor in communication with the detecting circuit and the memory selects one of the stored response characteristics matching the response characteristic measured by the detecting circuit. In some cases, an appropriate equalization is loaded based on the selected response characteristic.

Abstract: Systems and methods are described for clamping a headset in a calibration system using a clamp system that includes a clamp, platform, and one or more spindles (e.g., cushion spindles) to minimize or eliminate issues associated with positioning of headsets. The clamp system comprises a mount having a receptacle. When a device is introduced to the mount the receptacle receives at least a portion of a device. The clamp system includes a clamp attached to the mount and having a first arm rotateably coupled to a second arm that controls the first arm between an open position and a closed position. A platform and at least one spindle are connected to the first arm. When the device is present in the receptacle and the first arm is in the closed position the spindle contacts the device and seats or secures the device in the receptacle.

Abstract: An audio device includes an internal sound producer, an earphone jack, a detection circuit, a volume adjustor, and an amplifier. The detection circuit detects a connection between the earphone jack and an external sound producer, generates a first state signal when the connection is established, generates a second state signal when no connection is established. The volume adjustor is preset with first predetermined volume levels and second predetermined volume levels, generates first adjustment signals based on the first predetermined volume levels in response to the first state signal, and generates a second adjustment signal based on the second predetermined volume levels in response to the second state signal. The amplifier amplifies audio signals and adjusts the volume of the amplified audio signals according to the adjustment signals, then sends the amplified audio signals to the internal sound producer and earphone jack.

Abstract: An apparatus comprises at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform: monitoring at least one indicator dependent on a transducer mechanical integration parameter; and determining a change in the at least one indicator.

Abstract: A method of adjusting a loudness of an audio signal may include receiving an electronic audio signal and using one or more processors to process at least one channel of the audio signal to determine a loudness of a portion of the audio signal. This processing may include processing the channel with a plurality of approximation filters that can approximate a plurality of auditory filters that further approximate a human hearing system. In addition, the method may include computing at least one gain based at least in part on the determined loudness to cause a loudness of the audio signal to remain substantially constant for a period of time. Moreover, the method may include applying the gain to the electronic audio signal.

Abstract: A loudspeaker system capable of simultaneously providing listeners with a listening environment not giving them a feeling of strangeness. The loudspeaker system (100) comprises drive motors (111 to 113, 121 to 123) for so driving at least first R-channel loudspeaker (101) that the current first listening position is substantially included in the perpendicularly bisecting plane of a line joining the intersecting point of the first R-channel speaker (101) and its directivity axis and the intersecting point of the second R-channel loudspeaker (120) and its directivity axis and so driving at least fourth L-channel loudspeaker (104) that the current second listening point is substantially included in the perpendicularly bisecting plane of the line joining the intersecting point of a third L-channel loudspeaker (103) and its directivity axis and the intersecting point of a fourth L-channel loudspeaker (104) and its directivity axis. The loudspeaker system (100) further comprises their motor driver section (130).

Abstract: A music reproducing system has a music reproducing unit and a transducer unit connected to the music reproducing unit. The transducer unit includes a transducer, a main sensor, and attachment-state detector. The music reproducing unit includes an information processing part and a detection controller.

Abstract: A test system for evaluates acoustic feedback characteristics of a listening device by changing the transfer function of the signal path from an acoustic output to an acoustic input of the listening device. A variable filter comprises an acoustic input, an acoustic output, and a control unit for changing a transfer function of the filter. First and/or second acoustic propagation elements propagate acoustic sound, the first acoustic propagation element acoustically connecting the acoustic output of the listening device with the acoustic input of the variable filter, the second acoustic propagation element acoustically connecting the acoustic output of the variable filter with the acoustic input of the listening device; wherein the system is configured to allow a determination of acoustic feedback or a level of acoustic feedback of the listening device at different frequencies.

Abstract: A multi-channel audio system having multiple loudspeakers is used to obtain information on the location of one or more independent noise sources within an area covered by the loudspeakers. Within the multi-channel audio system, an audio output device has an input for coupling to and receiving audio signals from one or more audio sources; an audio processing module for generating a audio drive signals and providing them on respective outputs to a number of loudspeakers. A sensing module has inputs connected to respective outputs of the audio processing module, for receiving signals corresponding to sound sensed by the loudspeakers. The sensing module includes a discriminator for discriminating between signals corresponding to the audio drive signals and sensed signals from an independent noise source within range of the loudspeakers. A position computation module determines a two or three dimensional position of each independent noise source sensed, relative to the loudspeakers.

Abstract: A general model is provided for predicting a loudspeaker preference rating, where the model's predicted loudspeaker preference rating is calculated based upon the sum of a plurality of weighted independent variables that statistically quantify amplitude deviations in a loudspeaker frequency response. The independent variables selected may be independent variables determined as maximizing the ability of a loudspeaker preference variable to predict a loudspeaker preference rating. A multiple regression analysis is performed to determine respective weights for the selected independent variables. The weighted independent variables are arranged into a linear relationship on which the loudspeaker preference variable depends.

Abstract: A home theater system, an audio output adjusting device of a home theater system and methods thereof allow a prescribed or three-dimension audio configuration state for at least one prescribed position even when the prescribed location and/or a configuration of the speakers in a home theater system are modified. The device can include a checker for analyzing at least one of volume and delay time of a test audio signal outputted from speakers and providing the analyzed information to a controller. The controller can control output level and timing of audio signals subsequently output from the speakers based on the analyzed information.