Abstract: An audio signal is processed by passing it through a speaker, the cone of which is mechanically connected to a moving sleeve. A coil is wrapped around the sleeve. There is a metal shaft displaced inside of the sleeve. The shaft is connected to a magnet. Thus, movement of the speaker cone causes the coil to move through a magnetic field which creates a voltage across the coil which is subtracted from the original audio signal to produce a processed output.

Abstract: One embodiment of the present invention sets forth a method for managing a power state of an audio device resident in a graphics processing unit. The method includes the steps of directing audio data originated from a client application via an audio path in an audio driver stack to the audio device, determining whether an active stream of audio data along the audio path is present in response to a notification of an attempt to shut down the graphics processing unit, and requesting a plug and play manager to disable the audio device, if no active stream of audio data is present along the audio path.

Abstract: A control signal is generated for mechanical loudspeaker protection, or for other signal pre-processing functions. The procedure contains the following steps: perform a non-linearity analysis based on current and voltage measurements; use the results of the non-linearity analysis, and the voltage and current measurements to control audio processing for the loudspeaker thereby to implement loudspeaker protection and/or acoustic signal processing.

Abstract: A method and apparatus in a sound-reproduction system, in which method an electrical calibration signal is formed, an audio signal is formed in the loudspeaker from the calibration signal, the response of the audio signal is measured and analysed, and the loudspeaker system is adjusted on the basis of the measurement results. The operator is permitted to made additional alterations to the settings of the loudspeaker system on the basis of the measurement performed, the effects of the alterations are calculated and displayed to the operator without additional measurements, and the additional settings are implemented in real time in the loudspeaker system.

Abstract: A sound device includes an input terminal for electronic signal input; a line out terminal adapted to output electronic signals input on the input terminal to an external area; a delay arrangement adapted to delay electronic signals input on the input terminal for a certain time; a main output arrangement adapted to output electronic signals delayed by the delay arrangement. The delay arrangement delays signals in order to accommodate a time difference between (1) electronic signals output from an external sound device connected with the line output terminal and (2) electronic signals output from the main output arrangement. The time difference between sound that is output from the sound device and the one that is output from an external sound device that is connected with line connection terminal can be eliminated. Thus, it can prevent sound quality decrease due to time difference from happening.

Abstract: A sound field in a vehicle can be further readily and optimally corrected without forcing a user to do troublesome work. As an embodiment of the present invention, a measurement sound emitted through a left speaker or a right speaker at a further position in a view from a driver's listening point that was assumed on the head touching surface of the headrest of the driver's seat or the passenger seat is picked up, with a microphone provided as buried on a front panel of a head unit to be attached to the almost center position between the driver's seat and the passenger seat in the vehicle. And the frequency characteristic of the above sound is corrected, based on a tendency that the frequency characteristic of an audio sound reaching from the left speaker or the right speaker to the microphone is almost approximate to the virtual frequency characteristic of an audio sound that will reach from the left speaker or the right speaker to the driver's listening point assumed on the head touching surface.

Abstract: A loudspeaker is provided for receiving an incoming electrical signal and transmitting an acoustical signal that is directional and has a substantially constant beamwidth over a wide frequency range. The loudspeaker may include a curved mounting plate that has curvature over a range of angles. The loudspeaker may include an array of speaker drivers coupled to the mounting plate. Each speaker driver may be driven by an electrical signal having a respective amplitude that is a function of the speaker driver's respective location on the mounting plate. The function may be a Legendre function. Alternatively, the loudspeaker may include a flat mounting plate. In this case, the respective electrical signal driving each speaker may have a phase delay that virtually positions the speaker onto a curved surface.

Abstract: A wireless electronic arrangement configured to connect to an earphone arrangement may include a signal generator to generate an electric signal to be sent over the earphone arrangement; a measuring arrangement to make at least one measurement of the electric signal to determine a value indicating a current acoustic impedance of the earphone arrangement; a control unit to compare the determined value with a stored value indicating an operational acoustic impedance stored on a memory unit and, based on a result of the comparison, to select an operational mode of the wireless electronic arrangement based on the determined state of the earphone arrangement.

Abstract: An audio electronic device includes a player, an output device, a detector, and an automatic turn-off module. The player is for reproducing audio data. The output device is for transforming the audio data to audible sound. The detector is for detecting positional shift of the output device. The automatic turn-off module is for turning off the player based on the positional shift. An automatic turn-off method is also disclosed.

Abstract: An sound reproducing apparatus is configured such that a correction amount is calculated on the basis of a masking model, determiner for determining a correction parameter correcting input signal so as to be natural for human ears to hear on the basis of the calculated correction amount is provided, and a characteristic of correction filter is changed on the basis of a result of determiner when inputting vehicle speed information.

Abstract: Disclosed are a sound quality display apparatus, a sound quality display method, a computer readable medium on which a sound quality display program is recorded, and a sound camera. The apparatus includes a sound detector detecting sound generated from a sound source, a background photographing unit photographing background where the sound source is positioned, a sound source signal generator creating a sound source signal as a sound signal at a sound source plane where the sound source is positioned by analyzing the sound signal detected by the sound detector, a sound quality data generator creating sound quality data at the sound source plane by processing of the sound source signal, and a display displaying sound quality image data created by overlaying image data of the background photographed by the background photographing unit with the sound quality data.

Abstract: A system for monitoring the operation of one or more devices is provided. The system includes a microphone acoustically coupled to one the monitored device. An analog-to-digital-converter samples the microphone and a processor examines the resultant digital signal for the occurrence of an abnormal event.

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: A laser device, finder of range(s), and an inclinometer are used to figure a sound producing device's position adjustment(s). Laser beam(s) of the laser device and the inclinometer characterize direction(s) of sound produced by the sound producing device. The laser beam(s) can be one or more colors or patterns each characterizing the general direction of sound produced by the sound producing device. A stadium can have many speaker housings each including a speaker, the laser, range finder and the inclinometer. Each speaker housing is mounted on, and remotely adjustable with respect to, an structural member as to the general direction of sound emitted from the speaker housing into the area for stadium seating. The inclinometer's output can be rendered by a monitor seen by an audio engineer who also uses the laser beam's direction to make adjustments to the position of the speaker housing relative to its structural member.

Abstract: A system for testing audio of a motherboard includes a recording fixture and a computer. The recording fixture includes a recording device and a soundproof device. The soundproof device is configured for soundproofing the audio device and the recording device from environmental noise. The computer includes an output/input module, a comparing module and a displaying module. The output/input module is configured for outputting a first sound signal generated by the computer to the audio device to cause the audio device to generate a sound and receiving the sound generated by the audio device and recorded by the recording device from the recording device and transforming the sound into a second sound signal. The comparing module is configured for comparing the first and second sound signals. The displaying module is configured for displaying the compared result.

Abstract: An automated process for equalizing an audio system and an apparatus for implementing the process. An audio system includes a microphone unit, for receiving the sound waves radiated from a plurality of speakers, acoustic measuring circuitry, for calculating frequency response measurements; a memory, for storing characteristic data of the loudspeaker units and further for storing the frequency response measurements; and equalization calculation circuitry, for calculating an equalization pattern responsive to the digital data and responsive to the characteristic data of the plurality of loudspeaker units. Also described is an automated equalizing system including a acoustic measuring circuitry including a microphone for measuring frequency response at a plurality of locations; a memory, for storing the frequency responses at the plurality of locations; and equalization calculation circuitry, for calculating, from the frequency responses, an optimized equalization pattern.

Abstract: The invention relates to a method and a system for adapting a loudspeaker to a specific listening position relative to the loudspeaker according to which method and system the acoustic power radiated by the loudspeaker is corrected by means of a correction filter inserted in the signal path through the loudspeaker, the response of said correction filter being determined by comparison between the a quantity characterising the radiated acoustic power measured at an actual listening position and a similar quantity measured at a reference listening position. According to a specific embodiment of the invention said characterising quantities are the radiation resistances measured at the actual listening position and the reference listening position respectively.

Abstract: A wireless audio amplification system for classrooms and the like includes a system that allows for advanced listening and learning audio tools. The invention combines radio frequency and infrared technologies into one integrated system. A microphone includes a radio frequency receiver and an infrared transceiver. The microphone transmits voice signals to an audio amplifier unit that includes a radio frequency transceiver and an infrared transceiver.

Abstract: A sound insulating device used to carry out audio testing of a mobile phone is provided. The sound insulating device includes a sound speaker, a sound receiver, a clamping platform clamping the mobile phone thereon and a sound insulating chamber. The sound insulating chamber receives the sound speaker and the sound receiver and the clamping platform therein. The sound insulating chamber is enclosed by several sound insulating boards. Each sound insulating board includes a sound absorbing layer, a cushion layer and an aluminium-alloy layer combined together. The cushion layer is sandwiched between the absorbing layer and the aluminium-alloy layer.

Abstract: A mobile electronic device and a sound playback method thereof are provided. The mobile electronic device includes a sensor, a speaker, and a controller coupled to the sensor and the speaker. The sensor detects whether the speaker is blocked or not. When the speaker is blocked, the controller multiplies a sound signal by a transfer function and then outputs the multiplied sound signal. The speaker plays the sound signal outputted by the controller. The transfer function changes the direction in which the speaker plays the sound signal.

Abstract: A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

Abstract: An automatic tunable earphone connected with an external audio device has an audio signal output unit, a loudspeaker, a microphone, a data memorizer and an arithmetic unit. The audio signal output unit outputs a testing audio signal into the loudspeaker for generating a testing sound which is transmitted into an ear canal. The microphone placed in the ear canal receives the sound that has past through the ear canal and outputs a contrastive audio signal into the data memorizer for restoring. The arithmetic unit compares the contrastive audio signal with the testing audio signal to find out the frequency that needs to be strengthened. The frequency is restored into the data memorizer. When the external audio device outputs a raw signal, the arithmetic unit process the raw signal according to the frequency restored and outputs the processed signal into the loudspeaker for making sound.

Abstract: This disclosure describes techniques of automatically identifying a direction of a speech source relative to an array of directional microphones using audio streams from some or all of the directional microphones. Whether the direction of the speech source is identified using audio streams from some of the directional microphones or from all of the directional microphones depends on whether using audio streams from a subgroup of the directional microphones or using audio streams from all of the directional microphones is more likely to correctly identify the direction of the speech source. Switching between using audio streams from some of the directional microphones and using audio streams from all of the directional microphones may occur automatically to best identify the direction of the speech source. A display screen at a remote venue may then display images having angles of view that are centered generally in the direction of the speech source.

Abstract: A method for performance measurement and optimization of sound systems using electroacoustic measurements and a sliding band integration curve. Nearfield and spatially and temporally averaged broadband farfield responses are measured, averaged over a distinct set of frequencies, level matched, and weighted using a frequency-dependent ratio. The two curves are then combined to produce a third curve. The results indicate system performance in a listening space that matches human sensory response and provides means to optimize the sound system for the listening space.

Abstract: A light-emission responder can respond to input of sound, realizes versatility of response, and can be applied to various use forms. A microphone (110) converts a sound wave into an electric signal. An amplifying section (111) amplifies the electric signal and outputs it to an AGC section (112). The AGC section (112) adjusts the amplitude of the electric signal output from the amplifying section (111). A filter section (113) outputs an electric signal in a frequency band f1 out of the amplitude-adjusted electric signal to a comparing section (114). The comparing section (114) compares the input electric signal with a reference signal. If the voltage of the electric signal passing through the filter section (113) is higher than the reference voltage, the comparing section (114) outputs a signal “H”. When a signal “H” is output from the comparing section (114), a drive section (115) allows a light-emitting element (120) to emit light.

Abstract: A reproducing apparatus includes: a connecting part which is connected to an audio signal output apparatus through the three pole terminal; a storage part which stores content data including first and second channel audio data; a reproducing part which reproduces the content data; a first converting part which converts the first channel audio data included in the content data reproduced by the reproducing part into the first channel audio signal; a second converting part which converts the second channel audio data included in the content data reproduced by the reproducing part into the second channel audio signal; and a measuring part which measures at least any one of an amount of sound leakage in the second output unit caused by outputting the first channel audio signal and an amount of sound leakage in the first output unit caused by outputting the second channel audio signal.

Abstract: A frequency-characteristic-acquisition device that inputs a time-stretched-pulse signal to a system to be measured and that acquires information about a frequency characteristic of the system on the basis of a signal output from the system is provided. The frequency-characteristic-acquisition device includes a control unit which performs control so that the time-stretched-pulse signal is expanded in a time-axis direction and output to the system, and an acquisition unit that analyzes the signal output from the system and that acquires the frequency-characteristic information.

Abstract: A diaphragm for outputting sound by vibrating has a hole of a predetermined shape that allows another member to extend therethrough. A sound output apparatus includes a first member, the diaphragm, and a second member. The first member extends through the hole of the diaphragm, and the first member is screwed into the second member.

Abstract: A system including at least one system microphone, a processor connected to computer readable memory and the at least one system microphone, a measurement microphone connected to the processor, at least one audio speaker, a signal generator connected to the processor and configured to produce signals from the at least one audio speaker for both the measurement microphone and the at least one system microphone to measure the response of the at least one audio speaker, and a device manager application executable from the computer readable memory and configured to perform the steps of calibrating the at least one audio speaker with the measurement microphone using the signals of the signal generator based on the response of the at least one audio speaker thereby creating at least one calibrated audio speaker and calibrating the at least one system microphone with the at least one calibrated audio speaker using the signals of the signal generator.

Abstract: An earpiece, an electronic device, and a communication device, capable of reducing the influence of leakage of sounds from a clearance between the earpiece and an ear not to need an addition of parts are provided. An earpiece (100) includes an electromechanical acoustic transducer (10) having a diaphragm and for generating a sound by vibrating the diaphragm in response to an electric signal, a case (11) on which a sound hole (12) is provided and in which the electromechanical acoustic transducer (10) is housed, and a holding member (13) for holding the electromechanical acoustic transducer (10) in the case (11). The electromechanical acoustic transducer (10) has a sound pressure-frequency characteristic that is compensated in advance such that the sound pressure-frequency characteristic is planarized in a condition that the generated sound is leaked from a user's ear (14) being covered with the case (11).

Abstract: A device and method for improving the performance of an electro-acoustic transducer. An acoustic test signal is generated through the electro-acoustic transducer. The acoustic test signal is measured at multiple points on an ambient surface around the electro-acoustic transducer to create measured acoustic data. Based on the measured acoustic data, an acoustic power frequency response of the electro-acoustic transducer is calculated. A correction impulse response for the electro-acoustic transducer is determined based on the acoustic power frequency response. A correction filter applies the correction impulse response to a sound signal input to generate a sound signal output for playback through the electro-acoustic transducer.

Abstract: Methods and system for predicting sound pressure and/or sound pressure level caused by a vehicle horn are provided. An acoustic model of the horn is generated through the use of inverse numerical acoustics and boundary element methods. Additionally, an acoustic model of the vehicle is generated using boundary element methods. By combining these acoustic models and using the acoustic model of the horn as input into the acoustic model of the vehicle, sound pressure and/or sound pressure level at points within the acoustic domain encompassing the acoustic models can be predicted using boundary elements methods.

Abstract: An apparatus and a method for testing a sound card are applicable for detecting whether a sound leakage problem occurs to a sounding interface of the sound card. The testing apparatus includes a switching circuit and a switching unit. The switching circuit is electrically connected to the sounding interface and a sound-receiving interface of the sound card respectively, and has a plurality of transfer paths. The switching circuit is used to receive a first and a second sound channel, and to connect the first and the second sound channel to corresponding transfer paths, so as to transmit the audio signal back to the sound-receiving interface. The switching unit is electrically connected to the sounding interface and the switching circuit, for receiving the first and the second sound channel. The switching unit alters the transfer paths in the switching circuit according to a first and/or a second switching audio signal.

Abstract: A control method to automatically adjust the volume of a sound transmitter based on the measurement and the computation of the acoustic statistics of the room where the sound is emitted. The system comprises at least one sensor, a calculator determining statistics of the signal collected by the at least one sensor and a controller using these statistics provided by the calculator to adjust the volume of the transmitter in the room.

Abstract: The invention provides a method for manufacturing array microphones. First, signal delays of a plurality of microphones are measured. The microphones are then categorized into a plurality of categories according to the signal delays. A plurality of array microphones are then assembled with a number of component microphones selected from the same categories.

Abstract: An acoustic apparatus includes a positive value conversion section configured to convert, into a positive value, a response signal obtained by collecting a test signal emitted from a speaker using a microphone; a detection section configured to detect a first transient response part that becomes a first mountain portion of the converted response signal; an estimation section configured to estimate a rise point of the converted response signal from at least N points that contain a peak position of the first transient response part or the vicinity thereof; and a computation section configured to compute a time delay of audio collected by the microphone based on the estimated rise point and on a timing at which the test signal is generated.

Abstract: A communication system has a communication controller for multiplexing and transmitting a power supply voltage and an audio signal to be played back by a plurality of channels, and a plurality of communication adapters which start in response to reception of the multiplexed/transmitted power supply voltage, and play back the audio signal using a playback device. The communication controller includes an audio data modulation circuit configured to modulate audio data by shifting, based on a first modulation ratio, a modulation carrier clock for modulating the audio data and a data transmission rate for transmitting the audio data in order to set a command transmission frequency bandwidth, and a command data modulation circuit configured to modulate command data by shifting, based on a second modulation ratio, a modulation carrier clock for modulating the command data and a data transmission rate for transmitting the command data in the command transmission frequency bandwidth.

Abstract: A calibration device for calibrating a dynamic pressure sensor includes a power/control box and a portable calibration head disposed in electrical communication with the power/control box. The portable calibration head may include a calibration head housing having a housing opening and a speaker provided in the calibration head housing and communicating with the housing opening. The power/control box may be configured to induce emission of an acoustic calibration signal from the speaker of the calibration head.

Abstract: A method of controlling a loudspeaker of an electronic device provides voice coil temperature protection. When a power supply for the electronic device is first activated, a binding step is performed in which the loudspeaker impedance is determined and a temperature (such as ambient temperature) is accurately measured. These binding step measurements are used during subsequent use of the loudspeaker, to make the temperature measurements (based on voice coil impedance) as accurate as possible.

Abstract: An apparatus comprising a housing, a speaker, and an amplifier. The speaker may be mounted within the housing. A concave portion may be implemented in the housing and may be configured to hold a transformer. An amplifier may be implemented within the housing and may be configured to receive power through a connection to the transformer.

Abstract: The presentation of visual feedback in an electronic entertainment system is disclosed. One disclosed embodiment relates to a method of providing user feedback in an electronic entertainment system, wherein the method comprises inviting an input from a user, receiving a user input via a hand-held remote input device, performing a comparison of the user input received to an expected input, assigning a rating to the user input received based upon the comparison to the expected input, and adjusting light emitted by one or more lights sources on the input device based upon the rating.

Abstract: A feedback calibration system and a method for controlling an electronic signal are disclosed. The feedback calibration system includes an input controller adapted to modify an input signal in response to a control signal and generate a modified input signal, a signal processing block including a signal analyzer, wherein the signal processing block is adapted to process the modified input signal to generate an output signal and the signal analyzer is adapted to detect an undesirable condition of the output signal and transmit a detection signal corresponding to the undesirable condition, a transfer function estimator adapted to model and transmit a transfer function estimate of the signal processing block in real-time in response to the detection signal, and a programmable device adapted to transmit the control signal to the input controller for modifying the input signal, wherein the control signal is based upon the transfer function estimate.

Abstract: Methods, digital systems, and computer readable media are provided for determining a gain reduction parameter level for loudspeaker equalization by determining a noise score, an equalization effectiveness score, and an equalization non-effectiveness score for a candidate gain reduction parameter level, determining a composite quality score using the three scores, and designing a compensating filter for the loudspeaker using the candidate gain reduction parameter level if the composite quality score is better than composite quality scores of all other candidate gain reduction parameter levels.

Abstract: The present invention provides an advanced adaptive predistortion linearization technique to dramatically reduce nonlinear distortion in power amplifiers over a very wide instantaneous bandwidth (up to 2 GHz) and over a wide range of amplifier types, input frequencies, signal types, amplitudes, temperature, and other environmental and signal conditions. In an embodiment of the invention, the predistortion linearization circuitry comprises (1) a higher-order polynomial model of an amplifier's gain and phase characteristics—higher than a third-order polynomial model; (2) an adaptive calibration technique; and (3) a heuristic calibration technique. The higher-order polynomial model is generated by introducing, for example, a plurality of multi-tone test signals with varying center frequency and spacing into the power amplifier. From the power amplifier's corresponding output, the nonlinearities are modeled by employing a higher-order curve fit to capture the irregularities in the nonlinear transfer function.

Abstract: A method is provided for creating a series of digital signal processing (DSP) filters to improve the transient response of a loudspeaker, wherein the loudspeaker is formed of multiple components. The method includes generally six steps. The first step involves identifying a substantially linear, time-invariant, and spatially-consistent loudspeaker mechanism causing transient response distortion. The second step involves characterizing the identified mechanism. The third step involves determining the characterized mechanism's two-port response. The fourth step involves establishing a target response for the characterized mechanism. The fifth step involves calculating an ideal filter to achieve the target response. The sixth step involves designing a cost-reduced filter based on the ideal filter.

Abstract: A method for adjusting frequency response curve of a speaker comprises these steps: testing sensitivity of the speaker in at least two types of hardware conditions and recording corresponding frequency response curves; selecting a frequency response curve that comes closest to falling within a predetermined range for selected frequency ranges; and adjusting the selected frequency response curve with a filter.

Abstract: A system automatically determines an equalizing filter characteristic for a communication system within a vehicle. The communication system includes a loudspeaker and a microphone or microphone array. The system transmits a predetermined test signal through the loudspeaker and receive the test signal through the microphone or microphone array. Based on the predetermined test signal and the received test signal, a transfer function is developed. The equalizing filter characteristic is then developed from the transfer function.

Abstract: Outputs of a laser and an inclinometer are used to figure adjustments to a sound producing device's position, where both the laser beam of the laser and an output from the inclinometer characterize a general direction of sound produced by the sound producing device. The laser beam can be one or more colors or patterns each characterizing the general direction of sound produced by the sound producing device. A stadium can have many speaker housings each including a speaker, the laser, and the inclinometer. Each speaker housing is mounted on, and remotely adjustable with respect to, an structural member as to general direction of sound emitted from the speaker housing into the area for stadium seating. The inclinometer's output can be rendered by a monitor seen by an audio engineer who also uses the laser beam's direction to make adjustments to the position of the speaker housing relative to its structural member.

Abstract: A sound device is provided with an input terminal for electronic signal input; a low pass filter adapted to filter electronic signals input to the input terminal and to output low frequency range electronic signals, a switch arrangement adapted to switch between full range mode and sub woofer mode, a power amplifier adapted to amplify (1) electronic signals input from the input terminal when the switch arrangement selects full range mode, and (2) electronic signals output from the low pass filter when the switch arrangement selects sub woofer mode, and a speaker adapted to be operated by electronic signals amplified by the power amplifier.

Abstract: A speaker determination device determines a polarity of a speaker connected to output terminals. Test sound is outputted to an acoustic space via the connected speaker and is collected by a microphone. A reference signal is generated based on the test signal. A polarity determining unit compares predetermined frequency range components of the microphone signal obtained by the microphone and the reference signal, and determines the polarity of the speaker, i.e., whether the speaker is connected in positive phase or in negative phase.