Abstract: A method for operating a distributed wireless audio system including several loudspeaker cabinets all of which can communicate with each other as part of a computer network. The method receives temperature data that is indicative of temperature of a first loudspeaker cabinet, which has a network master responsibility of obtaining an audio signal from an audio source and wirelessly transmitting some of the audio signal to a second loudspeaker cabinet of several loudspeaker cabinets, for playback by the second loudspeaker cabinet, while playing back some of the audio signal by the first loudspeaker cabinet. The method determines whether a thermal threshold of the first loudspeaker cabinet has been reached, based on the temperature data. The method, in response to the thermal threshold being reached, gives up the network master responsibility from the first loudspeaker cabinet to the second loudspeaker cabinet, where doing so reduces temperature in the first loudspeaker cabinet.

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 microphone that includes a calibration signal generator that generates a calibration signal based upon a received invocation input. The calibration signal has known audio properties. The microphone also includes a transducer that converts sound energy into an electrical signal. An integrated circuit receives the electrical signal from the transducer. An external microphone interface provides the calibration signal to a device external to the microphone.

Abstract: Digital filters for processing an input audio signal for extending the audio capabilities and reducing distortion of a loudspeaker. Such extension may include a general sound pressure level (SPL) extension or a more targeted low frequency (LF) extension. This may be accomplished by modifying an original frequency response of the loudspeaker to achieve a particular extension, and applying a digital linear filter based on the modified frequency response to the audio signal. A nonlinear digital filter may also be applied to the audio signal reduce nonlinear distortions of the loudspeaker. The nonlinear digital filter may be based on an inverse of an electro-mechanical model of the loudspeaker. In this manner, a loudspeaker may be driven close to its maximum theoretical power capability without increasing distortion in the loudspeaker output.

Abstract: Method and apparatus for physical detecting movement of a playback device which plays audio sound via a speaker enclosed in the playback device. While the speaker plays the audio sound, a determination is made whether the playback device is physically moving. In response to the playback device physically moving, the audio sound played by the speaker is modified, for example, by adjusting the volume or equalization of the audio sound, so as to reduce the physical movement of the playback device.

Abstract: In accordance with embodiments of the present disclosure, a controller configured to be coupled to an audio transducer, may be further configured to receive an audio input signal, calculate a displacement compensation signal in a displacement domain of the audio transducer based on the audio input signal, convert the displacement compensation signal from the displacement domain into a voltage compensation signal in a voltage domain, and apply the voltage compensation signal to the audio input signal, or a derivative thereof, to minimize overexcursion of the audio transducer.

Abstract: An error model-based multi-zone sound reproduction method includes arranging a speaker array, and setting control points for a bright zone and a dark zone. The bright zone is a zone requiring the generation of an independent sound source. The dark zone is a zone not requiring the generation of an independent sound source. The method further includes conducting probability distribution modeling on the speaker frequency response errors. The method further includes, according to the error distribution model, respectively listing expected average sound energy expressions of the bright zone and the dark zone and a frequency response consistency constraint expression of the bright zone. The method further includes calculating a time-domain impulse response filter signal of each channel according to the time-domain sound energy contrast control criterion of the frequency response consistency constraint.

Abstract: A multi-mode speech communication system is described that has different operating modes for different speech applications. A signal processing module is in communication with the speech applications and includes an input processing module and an output processing module. The input processing module processes microphone input signals to produce a set user input signals for each speech application that are limited to currently active system users for that speech application. The output processing module processes application output communications from the speech applications to produce loudspeaker output signals to the system users, wherein for each different speech application, the loudspeaker output signals are directed only to system users currently active in that speech application.

Abstract: In one embodiment, a computer-program product embodied in a non-transitory computer readable medium that is programmed for performing an automatic speaker setup is provided. The computer-program product includes instructions for wirelessly receiving, via Near Field Communication (NFC), speaker information for at least one speaker in a speaker system. The computer-program product further includes instructions wirelessly transmitting the speaker information to an audio control unit that transmits audio data to the at least one speaker based on the speaker information.

Abstract: A device includes a delay element that can produce from the demodulated output of a radio receiver, a delayed signal that is delayed by an adjustable amount. A processor is coupled to the delay element and to a correlation element. The correlation element can correlate an acoustic signal from a transducer with information from the delayed signal from the delay element. The processor can adjust the delay and detect from the correlation element one or more peak correlations. The delays dictated by the processor, that produce one or more peak correlations, are used to deduce a use of a portable electronic device proximal to an operator's station, and to operate a switching element for altering services at the portable electronic device.

Abstract: A speaker system includes an electronic signal processing unit configured to split an input audio signal into a low-pass audio component and a high-pass audio component. The electrical signal processing unit includes a multi-rate finite impulse response (FIR) filter configured to downsample the low-pass audio component and sample the downsampled low-pass audio component at a first sampling rate lower than a second sampling rate used to sample the high-pass audio component. The speaker system further includes an electronic adaptive equalizer coefficient (EAEC) module configured to dynamically determine at least one equalizer coefficient based on a signal characteristic of the input audio signal. An electronic filter coefficient update unit (FCUU) is included and is configured to control a frequency response of the speaker system based on the at least one equalizer coefficient.

Abstract: A speaker driver comprising an amplifier, configured to receive a test signal that comprises a plurality of equivalent test-blocks, and provide measurement-signalling for a speaker at the amplifier output. The measurement-signalling comprising a plurality of measurement-blocks, wherein each of the measurement-blocks corresponds to the output of the amplifier for one of the plurality of test-blocks. The speaker driver also includes an output-current-sensor configured to: measure a current level of the measurement-signalling, and provide sensed-signalling that comprises a plurality of sensed-blocks, wherein each of the plurality of sensed-blocks corresponds to one of the plurality of measurement-blocks of the measurement-signalling.

Abstract: This application relates to methods and apparatus for loudspeaker protection. A loudspeaker protection system (100) receives a digital audio signal comprising a plurality of samples at an input node (IN). A delay block (15) delays the digital audio signal and a gain block (14) applies a controlled gain to the delayed digital audio signal. An excursion predictor (12) 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 (23) controls a gain setting (g) 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 (23) is configured to determine at least one gain setting {ga gi} to be applied to a set of samples {Va . . .

Abstract: Method of throttling a pilot tone for thermal monitoring of an electro-mechanical actuator starts by computing a power estimate based on a driving signal. A first temperature estimate is then computed based on the power estimate. The pilot tone may be generated by adjusting a level of the pilot tone based on at least one of the power estimate or the first temperature estimate. The pilot tone is injected into the driving signal to generate a driving output signal that is outputted by an electro-mechanical actuator. Other embodiments are also described.

Abstract: The present invention relates to a device for controlling a loudspeaker (14) in an enclosure, comprising: an input for an audio signal (Saudio_ref) to be reproduced; an output for supplying an excitation signal from the loudspeaker; means (26, 36, 38, 70, 80, 90) for calculating the excitation signal of the loudspeaker (14) at each moment based on the audio signal (Saudio_ref).

Abstract: A circuit and method for determining an ambient temperature for an electronic device having a loudspeaker are described. An amplifier drives the loudspeaker and first circuitry is configure to determine an overall temperature of the loudspeaker or the amplifier. Second circuitry is configured to determine a change in temperature of the loudspeaker or the amplifier resulting from power dissipated in the loudspeaker or the amplifier. Third circuitry is configured to subtract a signal representative of the change in temperature from a signal representative of the overall temperature and output a signal representative of the ambient temperature for the electronic device.

Abstract: A distortion compensator 10 acquires an asymmetric component included in a 1-bit pulse train outputted from a DSM 6 on the basis of an analog signal as an output signal obtained from the 1-bit pulse train, and an IQ signal as an input signal to be inputted to the DSM 6, and performs distortion compensation on the basis of the asymmetric component. The distortion compensator 10 is caused to store therein asymmetric component data representing the acquired asymmetric component. When acquiring the asymmetric component, the distortion compensator 10 acquires, as an asymmetric component, a difference between an output baseband signal obtained by orthogonally demodulating the analog signal as the output signal, and an input baseband signal before being orthogonally modulated.

Abstract: A nonlinear control system includes a controller, a model updater, and a model. The controller is configured to accept one or more input signals, and one or more updates generated by the model updater to produce one or more control signals. The system is configured to drive one or more transducers with the control signals to produce a rendered audio stream therefrom. The model updater is configured to analysis one or more portions of the audio stream and to update one or more aspects of the controller so as to alter performance of the transducer.

Abstract: A bit-by-bit error correction technique is disclosed that divides each bit transmission into an acknowledgment phase, an error correction phase, and a transmission phase.

Abstract: There is provided a system and method for playing a venue-specific object-based audio in a venue, the system comprising a memory, and a processor configured to receive an object-based audio including a plurality of audio components and create a venue-specific object-based audio based on a modification metadata by adjusting a level of at least one of the plurality of audio components of the object-based audio, the processor executing the object-based audio rendering software to render the venue-specific object-based audio in the venue.

Abstract: Example techniques may involve performing aspects of a spectral calibration using an applied spatial calibration. An example implementation may include receiving data representing spatial filters that correspond to respective playback configurations. The implementation may also involve causing the audio drivers to output calibration audio that is divided into a repeating set of frames, the set of frames including a respective frame for each playback configuration. Causing the audio drivers to output the calibration audio may involve causing an audio stage to apply, during each frame, the spatial filter corresponding to the respective playback configuration. The implementation may also include receiving data representing spectral filters that correspond to respective playback configurations, the spectral filters based on the calibration audio output by the audio drivers.

Abstract: A corrector is configured to transform audio signals to compensate for unwanted distortion characteristics of a loudspeaker. A tuning filter is configured to transform audio signals to incorporate desired distortion characteristics associated with a target loudspeaker. By chaining together the tuning filter and the corrector, an audio signal can be modified so that the loudspeaker, when outputting the audio signal, has response characteristics of the target loudspeaker.

Abstract: A special sound producing device which is loaded in a vehicle, and provides information to a vehicle crew by a special sound, includes a plurality of speakers which become the power sources of the special sound, and a control unit which generates frequency signals to make the special sound output from the plurality of speakers respectively, wherein when one information to be transmitted to the vehicle crew is transmitted by the special sound from the plurality of speakers, the control unit makes the frequencies of the frequency signals, the sound pressures of the special sound or the output timings to the speakers of the frequency signals to be different among the plurality of speakers.

Abstract: An optimised method for driving a loudspeaker is used for protecting the loudspeaker from damage due to excessive excursion or from overheating. The playback power of an incoming audio data stream is compared with a feedback power derived from the loudspeaker actuator and the comparison is used to adjust the frequency response of the loudspeaker, across individual sub bands.

Abstract: Problem to be Solved To effectively operate a microphone array provided with a sound source position estimating function. Solution A microphone control system of the present invention is realized by the use of a tablet-type computer, for example. On the display of the control system, a two-dimensional simulation diagram 302, for example, simulating the space in which the microphone array is disposed. An array mark 304 simulating the exterior shape of the microphone array is displayed on the simulation diagram 302. Further, a sound source mark 316 indicating the position of the sound source estimated by the microphone array is displayed on the simulation diagram 302. Thus, an operator can intuitively find the position of the sound source in the space in which the microphone array is disposed, by referring to the simulation diagram 302, the array mark 304 and the sound source mark 316 displayed on the simulation diagram. Thus, the usage of the microphone array is improved.

Abstract: In one embodiment, a pre-correcting engine generates a pre-corrected signal that may be used to mitigate air path distortions in sounds generated by loudspeaker systems that include loudspeakers coupled to waveguides. In operation, the pre-correcting engine virtually propagates a target signal in a reverse direction from the output of the waveguide to the output of a driver. Notably, the pre-correcting engine implements a propagation distortion compensation model that, when applied to the target signal, corrects for the effects of air path distortions on the target signal. The pre-correcting engine then transmits the pre-corrected signal to the driver. As the driver drives the waveguide based on the pre-corrected signal, effects of the pre-corrected distortions negate effects of air path distortions and the loudspeaker system generates high-fidelity sounds.

Abstract: This application relates to methods and apparatus for thermal protection of a loudspeaker (102). A controller (106) is configured to generate a control signal (Gmod) for modulating the gain of a signal processing chain (101) that drives the loudspeaker. The controller is configured such that the control signal is generated as a function of each of: an indication of voice coil temperature of the loudspeaker (TVC); an indication of power dissipation in the voice coil of the loudspeaker (PVC); and an indication of a reservoir temperature (Tres) of a thermal reservoir for heat flow from the voice coil. The reservoir temperature may be an ambient temperature of the environment or a temperature of a component of the loudspeaker or apparatus that acts as a heat sink.

Abstract: This accessory is apparatus capable of implementing a method that can affect one or more features of a portable electronic device when brought into a vehicle having a radio with a right and left speaker. The accessory has a sender for initiating transmission of an RF test signal at the portable electronic device. The RF test signal is arranged to cause the radio to produce a distinct pair of audio responses from the right and left speaker. The accessory has a discriminator for determining for the right and left speaker the corresponding return delay between transmission of the RF test signal and arrival of the distinct pair of audio responses at the portable electronic device.

Abstract: A system for receiving multiple conversations or messages and for playing the multiple conversations or messages with a mobile device and wireless earpieces. The system may determine various presentation parameters based on various characteristics of the received messages and may play the messages such that audio appears at distinguishing locations around the user. The system may change how messages and/or conversations are played in response to recognize a change in the focus of the user based on detected user inputs, such as body movement gestures.

Abstract: A control circuit configured to control a speaker is disclosed. The control circuit is configured to drive the speaker at a predetermined frequency that may correspond to a resonance frequency. The control circuit comprises a speaker driver in communication with a controller. The controller is operable to control a drive frequency of the speaker by monitoring at least a sample of each of a plurality of wavelengths of a current draw of the speaker in the form of a voltage signal. Based on the voltage signal, the controller is operable to identify a voltage differential and adjust the drive frequency of the speaker in response to the voltage differential to maintain the predetermined frequency.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal.

Abstract: The present application discloses an audio mixing method and system for audio mixing on original sound signals. The method includes: arranging a plurality of loudspeaker boxes according to predetermined positions to form a predetermined acoustic space, the predetermined acoustic space including a plurality of predetermined acoustic positions; and arranging, at predetermined acoustic positions in the predetermined acoustic space, sound track elements of each sound track among one or more sound tracks based on a predetermined rule. The present application provides an audio mixing method and system that implement subtle acoustic effects and provide better user experience.

Abstract: Examples described herein involve calibrating a playback device for a playback environment based on audio signals detected by a microphone of a network device as the network device moves about the playback environment. While the playback device is playing a first audio signal and the network device is moving within the playback environment from a first physical location to a second physical location, the network device may detect by a microphone of the network device, a second audio signal. The network device may then identify an audio processing algorithm based on data indicating the second audio signal, and transmit to the playback device, data indicating the identified audio processing algorithm. Similar functions may also be performed by the playback device being calibrated or a computing device, such as a server to coordinate calibration of the playback device.

Abstract: The volume of a sound generated from a source is monitored. Contextual data is collected from sensors. Based on the collected contextual data listeners of the sound are identified. The identified listeners are grouped into listeners that are bothered by the sound and listeners that are unbothered by the sound. An acceptable range for the volume of the sound is determined based on the bothered listeners and the unbothered listeners.

Abstract: A close-type speaker leak test system includes a first chamber, a second chamber, a third chamber, a fourth chamber, a vacuum generator, and a differential pressure gauge. Firstly, a negative pressure value of the first chamber and the second chamber is generated by the vacuum generator. A standard close-type speaker is placed within the third chamber, and the first chamber and the third chamber are in communication with each other. An under-test close-type speaker is placed within a fourth chamber, and the second chamber and the fourth chamber are in communication with each other. A pressure difference value between the third chamber and the fourth chamber is measured by the differential pressure gauge. If the pressure difference value is larger than a predetermined value, the under-test close-type speaker has the leak. Consequently, the misjudgment is reduced, and the testing efficiency is enhanced.

Abstract: A display device for a motor vehicle includes: a display for displaying vehicle information; a computing device receiving vehicle data via a data bus, evaluating the received vehicle data, and causing an acoustic warning as a function of the evaluation of the vehicle data; and a connection between the display device and a loudspeaker for outputting of the acoustic warning, the connection to the loudspeaker being connected to an input of an evaluation device in such a way that, depending on a signal present at the connection of the loudspeaker, a malfunctioning or a functioning of the acoustic output is determined, and an optical warning display is provided for outputting an optical warning in the case of a malfunction of the acoustic output.

Abstract: The present technology relates to a sound processing apparatus and method, and a program for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

Abstract: An audio output device is described comprising a speaker; an audio output circuit configured to receive a first audio signal and configured to supply the first audio signal and a second audio signal to the speaker, wherein the second audio signal comprises a higher frequency than the first audio signal; a microphone configured to receive an acoustic signal from the speaker in response to the first audio signal and the second audio signal and to convert the acoustic signal into a received audio signal and a determiner configured to determine a phase of a frequency component of the received audio signal corresponding to the second audio signal and to determine an excursion of the speaker by the first audio signal based on the phase.

Abstract: In one embodiment of the present invention, a loudspeaker parameter estimation subsystem efficiently and accurately estimates parameter values for a lumped parameter model (LPM) of a loudspeaker. In operation, the loudspeaker parameter estimation subsystem trains a neural network model based on responses generated via the lumped parameter model and the corresponding sets of parameter values. Subsequently, based on the relationship between the measured output response of a loudspeaker to an input stimulus, the loudspeaker parameter estimation subsystem estimates parameter values for the LPM of the loudspeaker. Advantageously, by sagaciously estimating parameter values for the LPM of loudspeakers, these NN-based techniques enable designers to leverage the LPM to reliably improve the design of loudspeakers, perform nonlinear correction of loudspeakers, and the like.

Abstract: Methods for modeling a loudspeaker having a passive radiator include applying a stimulus signal to a speaker within the cabinet, wherein the stimulus is applied over a frequency range. The sound pressure level (SPL) in the cabinet is measured as a function of frequency during application of the stimulus signal. At least one coefficient based on the measured SPL is derived, wherein at least one passive radiator parameter is a function of the at least one coefficient.

Abstract: Provided is an information processing system including a recognizing unit configured to recognize a first target and a second target on the basis of signals detected by a plurality of sensors arranged around a specific user, an identifying unit configured to identify the first target and the second target recognized by the recognizing unit, an estimating unit configured to estimate a position of the specific user in accordance with the a signal detected by any one of the plurality of sensors, and a signal processing unit configured to process each of signals acquired from sensors around the first target and the second target identified by the identifying unit in a manner that, when being output from a plurality of actuators arranged around the specific user, the signals are localized near the position of the specific user estimated by the estimating unit.

Abstract: Embodiments of systems and methods are described for inverting transformations of signals due to room acoustics. In some implementations, a transformation of a calibration signal from a particular location in a room may be determined. From this transformation, an inverse transformation may be determined and the inverse transformation may be applied to a speech signal received from a similar location.

Abstract: An audio system is described including an audio processor, an amplifier and a speaker and a distortion estimator. The distortion estimator calculates at least one of an expected response of the amplifier to an audio signal and an expected response of the loudspeaker to an audio signal. The distortion estimator is operable to generate a distortion prediction signal determined by a difference between an expected non-linear response of at least one of the loudspeaker and the amplifier and an expected linear response of at least one of the loudspeaker and the amplifier. A controller coupled to the audio processor and a control input of the audio processor may vary the operating parameters of the audio processor depending on the estimated distortion.

Abstract: A method of monitoring speaker temperature for speaker protection starts by generating a low level inaudible noise signal and injecting the low level inaudible noise signal in an audio signal. The voice coil resistance estimate that estimates a resistance of a voice coil of a speaker is then computed. The voice coil resistance estimate changes while the speaker is being driven by the audio signal that includes the low level inaudible noise signal. The temperature estimate is then computed based on the voice coil resistance estimate. The level of the audio signal may be adjusted based on the temperature estimate. Other embodiments are also described.

Abstract: In at least one embodiment, an apparatus for performing an audio measurement is provided. The apparatus includes a controller that is programmed to generate an audio test signal based on a first frequency range that comprises a first noise spectrum, a second frequency range that comprises a second noise spectrum, and a third frequency range that comprises a third noise spectrum and to transmit the audio test signal to an audio device that generates an audio response signal responsive to the audio test signal. The controller is further programmed to receive the audio response signal and to determine audio related characteristics for the audio device in response to the audio response signal.

Abstract: An audio output apparatus and a method are provided, which includes calculating a temperature value of a voice coil of the audio output apparatus using a heat transfer model algorithm, in response to power being supplied to the voice coil; adjusting an output level of an audio signal by determining a gain value to adjust the output level of the audio signal according to the calculated voice coil temperature value, and outputting the audio signal with the adjusted output level.

Abstract: A sound reproduction device includes a modulator having an output terminal for outputting a modulated carrier wave signal obtained by modulating a carrier wave signal in a ultrasonic band with an audible sound signal, a super-directivity loudspeaker connected to the output terminal, a capacitor connected between a ultrasonic wave source and a ground, first and second current detectors for detecting currents flowing through the super-directivity loudspeaker and the capacitor, a high-pass filter for outputting a filtered signal obtained by eliminating a low-frequency band component of the current detected by the first current detector, and a differential amplifier unit for outputting a signal corresponding to a difference between the filtered signal and the current detected by the second current detector. The ultrasonic wave source is configured to output the carrier wave signal such that the signal output from the differential amplifier unit is constant.

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

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal.

Abstract: A method for optimizing the design and sound field control of a compact loud-speaker array, which includes a plurality of loudspeakers located on a closed loudspeaker surface and the control of the emitted sound field by the loudspeakers within a limited reproduction subspace, having the steps of capturing the sound field using a plurality of microphones and adjusting filter coefficients that modify the alimentation signals of the loudspeakers to minimize the difference between reproduced signals captured by the microphones and target signals describing a target sound field. A conical reproduction surface encloses a reproduction subspace is defined such that the apex of the conical reproduction surface is within the closed loudspeaker surface. Loud-speakers are positioned on a limited loudspeaker surface and the closed loudspeaker surface.