Abstract: A temperature detecting and controlling integration device for the micro speaker is provided. After the filter receives an input signal, the power amplifier adjusts the power amplification, and the multi-frequency detection signal is generated with the waveform generator. The extracted signal is generated to drive the micro speaker to emit a sound signal. Afterwards, the voltage signals are extracted at two ends of the coil and the temperature signal is obtained by converting, capturing, and integrating to pass the temperature value to the external device, and the temperature value of the non-linear temperature-controlling unit is analyzed to adjust the compensation gain in real time. The smoothly control of speaker temperature and stable playback of the sound signals is played that can be achieved.

Abstract: A signal processing device includes a switching receptor, a storage, and a signal processor. The switching receptor receives switching of a speaker serving as a supply destination of a signal. The storage stores an optimal setting, which is obtained by measuring characteristics of the speaker selected by the switching, in association with the switching of the speaker. The signal processor reads out the optimal setting, which is associated with the switching received by the switching receptor, from the storage, and uses the optimal setting to process the signal to be supplied to the speaker.

Abstract: Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, determining how much to attenuate the audio signal when a power level of the audio signal exceeds a threshold power level, combining the audio signal with an auxiliary attenuation signal from an auxiliary attenuation source and a compressed attenuation signal from the feedback compressor circuit to create a combination signal, and generating an audio output signal of the feedback compressor circuit based on the combination signal.

Abstract: An embodiment of an audio controller device includes technology to control an output of a first audio signal from an audio output device in a human interface frequency range, where the human interface frequency range includes frequencies audible to humans, and control an output of a second audio signal from the audio output device in a device interface frequency range independent from the output of the first audio signal, where the device interface frequency range includes frequencies above an upper limit of the human interface frequency range. Other embodiments are disclosed and claimed.

Abstract: The present disclosure relates to the technical field of electronic devices, and discloses a method and an apparatus for controlling motor vibration. The method includes: obtaining an input signal based on an expected motor vibration curve, inputting a digital signal sequence of the input signal to an equalizer, and obtaining an output signal after processing by the equalizer, and inputting the output signal to a motor to control vibration of the motor, where the equalizer is a digital filter constructed based on a damping factor ? and a resonance frequency ?n of the motor, and a preset system sampling frequency fs, a preset damping factor ?d, and a preset cut-off frequency ?d. The method and apparatus provided in embodiments of the present disclosure have an advantage that an actual vibration effect of the motor can be consistent with an expected motor vibration curve.

Abstract: A measuring apparatus for measuring the force factor of a dynamic loudspeaker driver comprises first and second terminals for connecting a dynamic loudspeaker driver to the measuring apparatus, an amplifier configured to generate a voltage, a voltage measuring device configured to measure the voltage present at the first terminal, a displacement measuring device configured to measure the displacement of the membrane of the dynamic loudspeaker driver with respect to the magnet, and an electric/electronic device connected between the amplifier and the first terminal. The measuring apparatus is configured to operate in a first mode of operation the amplifier to generate a voltage and to operate the electric/electronic device to have a low impedance during the first mode of operation, such that the voltage is substantially present at the first terminal in order to displace the membrane with respect to the magnet.

Abstract: Attenuation of output levels is suppressed while interference due to a plurality of vibrators is reduced. An acoustic control device 30 performs correction processing of correcting phase delay characteristics including transmission systems from exciters 21L and 21R which are first and second vibrators connected with a rigid body, on acoustic signals SR and SL. Thereafter, the acoustic control device 30 controls the exciters 21L and 21R on the basis of the corrected acoustic signals SL1 and SR1.

Abstract: One embodiment provides a method comprising determining a potential energy in a loudspeaker, a kinetic energy in the loudspeaker, and an electrical energy in the loudspeaker based on a physical model of the loudspeaker. The method further comprises determining a total energy stored in the loudspeaker based on the potential energy, the kinetic energy, and the electrical energy. The method further comprises determining a maximum potential displacement of a diaphragm of a speaker driver of the loudspeaker based on the total energy, and limiting the total energy stored in the loudspeaker by attenuating a source signal for reproduction via the loudspeaker. An actual displacement of the diaphragm during the reproduction of the source signal is controlled based on the attenuated source signal.

Abstract: Examples described herein involve calibrating one or more playback devices. An example implementation involves a mobile device receiving data representing an audio signal recorded via a microphone at multiple locations between a first physical location and a second physical location within a given environment while one or more playback devices played back calibration audio within the given environment. Based on the received data representing the audio signal, the mobile device determines acoustic characteristics of the given environment. Based on the determined acoustic characteristics of the given environment, the mobile device determines one or more audio processing algorithms to adjust audio output of the one or more playback devices in the given environment to have a pre-determined audio characteristic and causes, via a network interface, the one or more playback devices to apply the determined one or more audio processing algorithms to playback of audio content.

Abstract: A method and apparatus of load detection for an audio amplifier system is described. A load detector includes a first load terminal and a second load terminal; a controller coupled to the first and second load terminals and configured to in a first control loop, vary a first current supplied to a first load terminal dependent on the difference between a first reference signal and the detected first load terminal voltage; and in a second control loop, vary a second current supplied to the second load terminal dependent on the difference between a second reference signal and the detected second load terminal voltage; and to determine a current through a load connected between the first load terminal and the second load terminal from the second current value, and a voltage across the load from the detected voltage difference between the first load terminal voltage and the second load terminal voltage.

Abstract: A terminal device and a method for removing dust in a sound outlet hole of a loudspeaker are provided. The method includes the following: measuring a natural frequency of the loudspeaker, generating an audio file for removing dust according to the natural frequency, and prestoring the audio file for removing dust, wherein a vibration amplitude of a vibration diaphragm of the loudspeaker is at a maximum when the vibration diaphragm is driven to vibrates by an exciting current having the natural frequency; reading the prestored audio file for removing dust; and controlling the loudspeaker to play the audio file for removing dust, to generate air flow in the sound outlet hole, wherein when the loudspeaker plays the audio file for removing dust, a wind speed of the air flow through the sound outlet hole is at a maximum.

Abstract: This application relates to methods and apparatus for loudspeaker protection. A loudspeaker protection system receives a digital audio signal comprising a plurality of samples at an input node. A delay block delays the digital audio signal and a gain block applies a controlled gain to the delayed digital audio signal. An excursion predictor is configured to receive a version of the audio signal from the signal path upstream of the delay block and determine a predicted excursion for a loudspeaker based on the audio signal. A gain controller controls a gain setting of the gain block in response to the predicted excursion and a first loudspeaker impulse response model and a predetermined excursion limit. The gain controller is configured to determine at least one gain setting {ga gi} to be applied to a set of samples {Va . . .

Abstract: A memory previously stores, as reference impedance characteristics, frequency characteristics of impedance of the speaker during a normal operating state of the speaker. During use of the speaker, a detection circuitry detects, as current impedance characteristics, frequency characteristics of impedance of the speaker on the basis of a real-time audio signal being supplied to the speaker. A determination circuitry determines presence/absence of an abnormality of the speaker on the basis of comparison between the detected current impedance characteristics and the stored reference impedance characteristics. Thus, it is possible to check the operation of the speaker based on of the real-time audio signal without using any dedicated test signal. In this way, it is possible to detect presence/absence of an abnormality of the speaker on the basis of a real-time audio signal being supplied to the speaker, for example, during an actual performance in a concert.

Abstract: A mobile terminal includes: a controller, and a piezoelectric excitation portion including a first excitation source, and a second excitation source surrounding the first excitation source, wherein: the controller is configured to control the first excitation source to generate a first vibration driving force and control the second excitation source to generate a second vibration driving force when the mobile terminal is in a handset mode, such that at a position where the second excitation source is located, a first vibration wave generated by the first vibration driving force and a second vibration wave generated by the second vibration driving force have opposite vibration directions, and a vibration amplitude of the second vibration wave is less than or equal to that of the first vibration wave.

Abstract: One embodiment provides a system for nonlinear control of a loudspeaker. The system comprises a current source amplifier connected to the loudspeaker, and a controller connected to the current source amplifier. The controller is configured to determine a target displacement of a diaphragm of the speaker driver based on a source signal for reproduction via the loudspeaker, determine a control current based on the target displacement of the diaphragm and a first physical model of the loudspeaker, and transmit a control current signal specifying the control current to the current source amplifier. The current source amplifier outputs the control current to drive the speaker driver based on the control current signal. An actual displacement of the diaphragm during the reproduction of the source signal is controlled based on the control current, via the control current signal.

Abstract: The various implementations described herein include methods, devices, and systems for automatic audio equalization. In one aspect, a method is performed at an electronic device that includes speakers, microphones, processors and memory. The electronic device outputs audio user content from the speakers and automatically equalizes subsequent audio output of the device without user input. The automatic equalization includes: (1) obtaining audio content signals, including receiving outputted audio content at each microphone; (2) determining from the audio content signals phase differences between microphones; (3) obtaining a feature vector based on the phase differences; (4) obtaining a frequency correction from a correction database based on the obtained feature vector; and (5) applying the obtained frequency correction to the subsequent audio output.

Abstract: A wireless speaker audio system configured to receive audio information wirelessly transmitted by an audio source including first and second wireless transceivers. The first wireless transceiver establishes a bidirectional secondary wireless link with the audio source for receiving and acknowledging receipt of the audio information. The first and second wireless transceivers communicate with each other via a primary wireless link. A wireless audio system including an audio source and first and second wireless transceivers. The first and second wireless transceivers communicate via a primary wireless link. The audio source communicates audio information to the first wireless transceiver via a secondary wireless link which is configured according to a standard wireless protocol. The first wireless transceiver is configured to acknowledge successful reception of audio information via the secondary wireless link. Other embodiments are also described and claimed.

Abstract: A computer is programmed to receive audio data from a first vehicle sensor following an instruction to a second vehicle to actuate sound output. The computer generates diagnostic data about the received audio data. The computer may then actuate output based on the diagnostic data.

Abstract: A headphone that may reduce acoustic noise from a tactile vibration driver includes a housing, and an acoustic driver and tactile vibration driver within the housing. The tactile vibration driver is configured to generate tactile vibration sufficient to be felt by a user responsive to the input signal. The headphone also includes a noise cancellation unit coupled with the acoustic driver, the noise cancellation unit configured to: generate an adjustment signal based, at least in part, on a transfer function associated with the tactile vibration driver generating acoustic noise incidental to the tactile vibrations; and adjust the input signal responsive to the adjustment signal to transmit an output signal for reproduction by the acoustic driver. Related methods for operating and making such headphones are also disclosed.

Abstract: User adjustment of an audio effect of an audio device is facilitated to match a hearing sensitivity of a user. The user can tune the audio device with a minimum perceptible level unique to the user. The audio device can adjust the audio effect in accordance with the minimum perceptible level. For example, a loudness level can adjust automatically to ensure that the user maintains a perceptible loudness, adjusting according to environmental noise and according to the minimum perceptible level. Also described herein are apparatuses, systems and methods related to an audio device equipped with embedded audio sensors that can maximize a voice quality while minimizing the effects of noise.

Abstract: One embodiment provides a device comprising a speaker driver, a microphone configured to obtain a measurement of a near-field sound pressure of the speaker driver, and a controller. The controller is configured to determine a velocity of a diaphragm of the speaker driver, and automatically calibrate sound power levels of audio reproduced by the speaker driver based on the velocity and the measurement of the near-field sound pressure to automatically adjust the sound power levels to an acoustic environment of the device.

Abstract: Examples described herein involve identifying one or more error conditions during calibration of one or more playback devices in a playback environment. A microphone of a network device may detect and sample an audio signal while the one or more playback devices in the playback environment plays a calibration tone. A processor of the network device may then receive, from the microphone, a stream of audio data. The audio data may include an audio signal component and a background noise component. As a subset of the audio data is received, the processor may identify based on the audio data, the one or more error conditions. The processor may then cause a graphical display to display a graphical representation associated with the identified error condition.

Abstract: A motor vehicle sound generator system has a housing in which a loudspeaker is accommodated. The housing has at least a first orifice and a second orifice. The orifices each have a sound transmission member provided thereon. The first orifice is arranged in a front side region of the loudspeaker and the second orifice is arranged in a rear side region of the loudspeaker as viewed in an axial direction of the loudspeaker. The sound generator system has a first sound outlet opening that is associated with the first orifice and a second sound outlet opening that is associated with the second orifice. The sound exits into open air through the first and second sound outlet openings, with the first and second sound outlet openings being free of exhaust gas flow.

Abstract: A quantitative optical microscopy arrangement is described. Specifically, a digital filter derived from linear discriminant analysis is described for recovering impulse responses in applications that may include photon counting from a high speed photodetector and applied to remove ringing distortions from impedance mismatch in multiphoton fluorescence microscopy. Training of the digital filter is achieved by defining temporally coincident and non-coincident transients and identifying the projection within filter-space that best separates the two classes. The training allows rapid data analysis by digital filtering. The LDA filter is also capable of recovering deconvolved impulses for single photon counting from highly distorted ringing waveforms from an impedance mismatched photomultiplier tube.

Abstract: A method clusters audio sources in virtual environments. The method is performed at a virtual-reality device displaying a virtual environment. The device identifies two audio sources in the virtual environment. For each of the two audio sources, the device determines a bounding box in the virtual environment. Each bounding box includes termination points for a respective plurality of rays emanating from a point in the virtual environment corresponding to the audio source. The device applies an overlap test to the bounding boxes to determine whether the two audio sources are in a same room. The device forms an angle according to rays from the location of the listener to the audio source points. When the two audio sources are in the same room and the angle is less than a predetermined threshold angle, the device clusters the two audio sources together, including rendering combined audio for the two audio sources.

Abstract: Various systems and methods are disclosed herein to increase the quality of the sound delivered to a user and allow personalization to optimize listening performance and comfort under atypical listening conditions, environment specific adjustment, and data capture to assist in the personalization of the system to the user's needs and preferences. Features disclosed include sound level rating systems that aggregate noise data detected by user's mobile phones or hearing devices to provide a database of real-time noise levels. Additionally, a user's sound settings may be saved in the system by location so that they may be recalled when re-entering a specific location. A remote clinician may tune a hearing device, or a user can tune the device using a pre-recorded audio sample. Also, a user may replay the last X seconds of audio recorded by their hearing device.

Abstract: Method and apparatus for automatic gain control utilizing sound source position information in a shared space having a plurality of microphones and a plurality of sound sources. Sound signals are received from the microphones. One or more processors locate position information corresponding to each of the sound sources. The processor(s) determine the distance to each of the sound sources from each of the microphones. The processor(s) define a predetermined gain weight adjustment for each of the microphones. The processor(s) apply the defined weight adjustments to the microphones to achieve a consistent volume of the desired plurality of sound sources. The processor(s) maintain a consistent ambient sound level regardless of the position of the sound sources and the applied gain weight adjustments. The processor(s) output a summed signal of the sound sources at a consistent volume with a constant ambient sound level across the plurality of sound source positions.

Abstract: A device having an earphone is provided to be attached to the tragus of the outer ear of a person, wherein two levers held to one another in a pivotal manner from opposite sides, driven by an elastically preloaded part, press onto the tragus. The levers both have two arms, wherein the pivot axis is at a distance from the ends of the two levers in the longitudinal direction of the lever and lies outside the normal projection towards the surface of the tragus when the device is attached as intended to the tragus, and/or the elastically preloaded part is a pre-curved bending spring which is pre-curved by more than 360°.

Abstract: A loudspeaker diaphragm state estimation method includes: adjusting a weight value of a diaphragm displacement model by adaptive filtering until an error between an estimated value of a driving voltage of a loudspeaker and a measured value of the driving voltage is less than a predetermined threshold; estimating a diaphragm relative displacement of the loudspeaker according to the diaphragm displacement model corresponding to a final determined weight value; determining a diaphragm relative speed at a next moment based on an input current, a product value of a vector determined by an estimated value of a diaphragm relative speed and an estimated value of a diaphragm relative displacement, and a weight value vector obtained at a present moment; and determining an estimated value of the driving voltage using the estimated value of the diaphragm relative speed, the input current, and a DC impedance of the loudspeaker at the present moment.

Abstract: A portable audio system includes a housing defining a common acoustic cavity a woofer disposed at a first end of the housing in the common acoustic cavity and a tweeter disposed at a second end of the housing, opposite to the first end, in the common acoustic cavity facing the woofer. A waveguide is disposed in the common acoustic cavity between the woofer and the tweeter to separate the woofer and the tweeter. The waveguide disperses and distributes sound waves generated by the woofer and tweeter.

Abstract: A speaker model may implement a direct voltage-to-excursion model in an adaptive filter for modeling the speaker without developing a first electrical-only model and then converting the model to a mechanical model. The voltage-to-excursion model may allow for modeling of different kinds of speakers, such as sealed, ported, or vented speakers. A transfer function may be developed in the adaptive filter for the voltage-to-excursion model, and that transfer function re-used for prediction of excursion values based on an audio signal. Speaker protection may be performed to take steps to prevent speaker damage when a predicted excursion value exceeds safe limits. The voltage-to-excursion model may operate in displacement or displacement-related domains (e.g., velocity and back emf).

Abstract: A predictive back-emf protection methodology for an electromechanical system, including a signal processor that processes a source signal to provide a modified source signal, a driver that converts the modified source signal to a drive signal, and an electromechanical transducer that generates, from the drive signal, a transducer response, and a back-emf signal coupled back to the driver output. A predictive back-emf generator (such as a routine in the signal processor) is characterized by a back-emf transfer function (linear parameterized model of the electromechanical transducer) for transforming an input signal into a transform back-emf representation of a back-emf signal predicted by the back-emf transfer function as a response of the electromechanical transducer to such input signal. The signal processor processes the source signal based on the transform back-emf representation to generate the modified source signal input to the driver.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a computing device having a controller to modify sound produced by a gaming application according to an otological profile of a user. Additional embodiments are disclosed.

Abstract: This application discloses various audio play control methods and apparatuses, and various electronic devices. One of the audio play control method comprises: determining play control information of a first audio that is being played or to be played by an electronic device, in response to that an acoustic part of the electronic device is blocked; and sending the play control information to play, according to the play control information, a corresponding audio through an acoustic part of at least another electronic device. This application can compensate for or replace an audio play sound effect of the electronic device through the at least another electronic device, thereby improving convenience when a user uses.

Abstract: A speaker, a television provided with the speaker and a multimedia device are provided. The speaker includes a box body and a plurality of transducer units installed within the box body by means of parallel connection: front sound cavities of the plurality of transducer units are independent of each other, respectively; and the plurality of transducer units share one back sound cavity. The speaker possesses such advantages that it has a high sound pressure level output and a smaller power consumption, it is usable to reduce the back-cavity acoustic resistance, and it is usable to raise the lower frequency limit in the low frequency range.

Abstract: A sound-reproducing method that includes the steps outlined below is provided. A playback sound is generated by applying original audio into a test environment. The playback sound is received to generate received sound data. At least one test environment spatial parameter corresponding to the test environment is calculated according to known audio data related to the original audio and the received sound data. Input audio is modified by applying the test environment spatial parameter thereto to generate reproduced audio.

Abstract: In various embodiments, a circuit arrangement includes a calibration filter set up to receive a signal based on a first signal and to provide a calibrated signal, the first signal being a signal which is provided by an analog/digital converter and is based on an analog signal, the analog signal being provided by at least one sensor of a sensor arrangement, a filter arrangement set up to receive a signal based on the first signal and to provide a second signal, and a controller set up to select a sensor-specific control signal dependent on the frequency response of the sensor from a plurality of control signals and to provide the calibration filter with said signal, the calibrated signal being based on the first signal and the sensor-specific control signal and corresponding or substantially corresponding to a predefined spectral mask in a predefined frequency range.

Abstract: A method and apparatus for controlling a volume of a device is provided herein. During operation, a first device will receive a volume-adjustment message from a second device, along with an identity of an active application being run on the second device. The volume-adjustment message indicates whether or not the second device increased or decreased their volume. The first device will increase or decrease their volume in a similar manner as the second device if the first device is currently running a same active application as the second device; otherwise the first device will adjust its volume in an opposite direction as indicated in the volume-adjustment message.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model, and associated parameters, of the loudspeaker as well as a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the voltage and current sense data, to generate the estimated impedance model that is converted to an excursion model of the loudspeaker. A downstream audio signal processing component, based on the excursion model, or parameters thereof, limits a predicted excursion of the loudspeaker utilizing excursion-constraining processing circuitry that includes a non-linear constraint filter. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

Abstract: A speaker driving apparatus can include: a sound pressure feedback unit configured to generate an electrical signal indicative of a sound pressure of a loudspeaker; a compensation unit configured to compensate an input audio signal according to the electrical signal indicative of the sound pressure of the loudspeaker to generate a compensated audio signal, where the sound pressure of the loudspeaker is linearly related to the input audio signal; and an amplification unit configured to amplify the compensated audio signal in order to drive the loudspeaker.

Abstract: Examples described herein involve calibrating a playback device. An example implementation receives, from a network microphone device (NMD), data indicating second audio signal detected by the NMD at multiple locations between a first physical location and a second physical location within a given environment while the network microphone device is moving from the first physical location to the second physical location, the second audio signal representing acoustic echo of a first audio signal played by a playback device. Based on the detected second audio signal, the implementation determines an audio characteristic of the given environment. Based on the determined audio characteristic, the implementation determines an audio processing algorithm to adjust audio output of the playback device in the given environment to have a pre-determined audio characteristic that is representative of desired audio playback qualities.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model, and associated parameters, of the loudspeaker as well as a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the voltage and current sense data, to generate the estimated impedance model of the loudspeaker by combining the fitted impedance components. The resulting estimated impedance model is converted to an excursion model of the loudspeaker. A downstream audio signal processing component utilizes the excursion model, or parameters thereof, to limit a predicted excursion of the loudspeaker. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

Abstract: A signal from a sensor may be received indicative of an acceleration of a moving component of a transducer at a location where the sensor is mounted. A position the moving component may be determined based on the acceleration. The position of the moving component may be compared with a reference to output a measure of distortion associated with the transducer. Nonlinearities in audio output by the transducer may be corrected based on the measure of distortion.

Abstract: One embodiment of the present invention sets forth a technique for calibrating an audio system. The technique includes transmitting information to a robotic vehicle for positioning a microphone at a plurality of different listening locations within a listening environment and, for each different listening location, acquiring a sound measurement via the microphone. The technique further includes calibrating at least one audio characteristic of the audio system based on the sound measurements acquired at the different listening locations.

Abstract: An audio system is described that corrects for linear and nonlinear distortions. The system can include a physical loudspeaker system responsive to an audio input signal, an adaptive circuit, e.g., with a recurrent neural network, to correct for non-linear distortions from the loudspeaker.

Abstract: An electronic apparatus of an audio output system is provided. The electronic apparatus includes: a first communicator configured to transmit a first radio signal and receive a first response signal; a second communicator configured to transmit a second radio signal and receive a second response signal; and a processor configured to: determine a first distance between the first communicator and an audio output apparatus based on the first response signal, determine a second distance between the second communicator and the audio output apparatus based on the second response signal, determine a location of the audio output apparatus based on the first distance and the second distance; establish a communication connection with the audio output apparatus based on one from among the first response signal and the second response signal; and set a channel of the audio output apparatus based on the determined location of the audio output apparatus.

Abstract: A hearing aid system and method of operating the hearing aid system, wherein the impedance of a hearing aid receiver is measured, values of two hearing aid receiver parameters are derived based on the measurements, an electro-acoustical model of the receiver is provided using the derived values, the model is used to predict (i) a non-distorted membrane displacement based on the derived values of the parameters measured at a zero bias voltage, and (ii) a distorted membrane displacement based on the derived values of the parameters measured at a non-zero bias voltage, based on the predicted displacements, a compensation gain is determined suitable to compensate non-linear distortion of the hearing aid receiver, and the compensation gain is applied to the processing of a hearing aid input signal.

Abstract: The invention provides a loudspeaker nonlinear compensation method. The method includes steps: obtaining system parameter of the loudspeaker, the No. i time-domain excitation voltage signal and the No. i state vector of the loudspeaker; compensating the No. i time-domain excitation voltage signal according to the system parameter and the No. i state vector and obtain i compensation voltage signal; obtaining the No. i+1 state vector according to the calculation of the system parameter and the No. i compensation voltage signal; outputting the No. i compensation voltage signal and record the quantity of the compensation voltage signal; judging whether the quantity of the compensation voltage signal is equal to the preset number value.

Abstract: An audio system has multiple loudspeaker devices to produce sound corresponding to different channels of a multi-channel audio signal such as a surround sound audio signal. The loudspeaker devices may have speech recognition capabilities. In response to a command spoken by a user, the loudspeaker devices automatically determine their positions and configure themselves to receive appropriate channels based on the positions. In order to determine the positions, a first of the loudspeaker devices analyzes sound representing the user command to determine the position of the first loudspeaker device relative to the user. The first loudspeaker also produces responsive speech indicating to the user that the loudspeaker devices have been or are being configured. The other loudspeaker devices analyze the sound representing the responsive speech to determine their positions relative to the first loudspeaker device and report their positions to the first loudspeaker device.

Abstract: Speakers do not always operate linearly. Linearity of the speaker can affect the quality of the sound produced by the speaker, i.e., causing distortions in the sound, if the nonlinearites are not accounted for. To determine nonlinearities of the speaker, the speaker is often modeled and measurements are made to estimate the characteristics of the speaker based on the model. By using an angle sensor and a light source, a speaker manager can make a direct measurement of excursion or displacement of the speaker. Moreover, when the angle sensor, the light source, and the light beam are configured appropriately with respect to the moving cone of the speaker, the measurement can be substantially linear with respect to the amount of excursion or displacement. Such measurements are far simpler to use and in some cases more accurate than measurements made by other types of systems.