Abstract: An apparatus and method of loudspeaker equalization. The method combines default tunings (generated based on a default listening environment) and room tunings (generated based on an end user listening environment) to result in combined tunings that account for differences between the end user listening environment and the default listening environment.

Abstract: Methods and set-top boxes are disclosed for automated error detection. The method includes detecting, on an electronic device, a display device connected to the electronic device via a cable; identifying, on the electronic device, a type of display device connected to the electronic device via the cable; classifying, on the electronic device, the type of display device connected to the electronic device via the cable as one of a specific television model or a regular television model; and executing, on the electronic device, a fix for the specific television model. The method can include detecting, on an electronic device, a HDMI cable-related error from a HDMI transmitter of the electronic device; classifying, on the electronic device, a cause of the HDMI cable-related error from the HDMI transmitter; and communicating, by the electronic device, a recommendation to a user to remedy the HDMI cable-related error detected by the HDMI transmitter.

Abstract: A feeder/interconnection/return (FIR) Cable Loop System vastly improves the distribution, equalization, efficiency and power delivery of high performance multiple speaker systems, without the complexities and inherent inefficiencies of existing systems. This is achieved by a feeder cable which connects from an amplifier to the first speaker, while interconnecting cables connect all the speakers to each other. A return cable connects the last speaker back to the amplifier. This results in balanced power, significant improvement in frequency response, and dynamic range.

Abstract: A calibration device and method of manufacturing the same. The calibration device for a hearing earpiece configured to at least partially protrude into the ear canal of a user includes a calibration base. The calibration base a calibration insert. An air chamber is defined between the calibration base and the calibration insert. The calibration device also includes at least one earpiece receiving mechanism defined on the calibration insert. The at least one earpiece receiving mechanism is configured to create an sealed connection between given ear piece and the air chamber for calibration. The earpiece receiving mechanism is configured to at least partially receive the given earpiece. A corresponding method of manufacturing is also included.

Abstract: In the present disclosure, a method comprising: receiving a first advertisement message for measuring a declination angle between devices; measuring each declination angle between a reference point set in the control device and positions of at least one device based on the first advertisement message, wherein the positions of the at least one device are respectively adjusted such that each declination angle satisfies a specific angle for receiving a voice signal, based on the measured respective declination angles; receiving each of a first voice signals based on the respective adjusted positions from the at least one device; and measuring a volume of the received first voice signal, respectively, wherein based on the measured volume, an audio signal output angle of the at least one device is adjusted such that an audio signal received by the control device is incident at right angles to the control device.

Abstract: An electronic device according to various embodiments, may include a speaker, a temperature sensor disposed in at least some area inside the speaker, and at least one processor configured to obtain a first temperature of the at least some area using the temperature sensor, obtain a second temperature based on measuring impedance of a voice coil included in the speaker, determine a temperature of the speaker based on at least one of the first temperature and the second temperature, and change a function of the electronic device based at least on the determined temperature of the speaker.

Abstract: Techniques, methods, systems, and other mechanisms for measuring the excursion of a speaker while being actively driven. Measuring excursion can involve attaching a flexible printed coil (FPC), including a sense coil, to the speaker, and monitoring an induced current as produced though the sense coil, and further detecting that violation of an excursion limit for the speaker may likely occur.

Abstract: Systems and methods for calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

Abstract: A method for controlling a loudspeaker having an electromechanical force transducer and a diaphragm by: Providing a non-linear electromechanical model configured to apply one or more desired conditions to a loudspeaker input digital audio signal, i.e. to an analogic input signal converted in a digital input signal; Providing an inverse non-linear electromechanical model of the transducer configured to receive a signal processed by the non-linear model and to linearize at least one mechanical and/or electrical and/or electromechanical non-linearity of the transducer; Converting the digital output signal of the electromechanical model into an analog signal for the transducer, So that the output signal comprises an input voltage signal for the transducer and at least the second non-linear model is a digital wave filter (hereinafter referred to as Wave Digital Filters, WDF) to provide a directly computable function in the discrete-time domain to get the input voltage signal for the transducer.

Abstract: A method of automatically setting an acoustic signal of a device for space monitoring by using the acoustic signal is proposed. The proposed is a technology for automatically setting the acoustic signal emitted from the device for space monitoring by using the acoustic signal for more precise measurement.

Abstract: According to an embodiment, an electronic device comprises a memory, a communication module, a first speaker including at least one vibration component, at least one first microphone, and a processor configured to output a first sound having a predetermined frequency via the first speaker when a closed space is formed with the electronic device mounted on a cradle, obtain a third sound, which is a reflection of the first sound in the closed space, via the at least one first microphone, obtain a fourth sound, which is a reflection of a second sound in the closed space, via the at least one first microphone, the second sound output from a second speaker included in an external electronic device located in the closed space, and identify whether the performance of the first speaker, the at least one first microphone, and the second speaker is normal, based on the third sound and the fourth sound.

Abstract: The invention relates to the technical field of microphones, in particular to an online trimmed MEMS microphone. The online trimmed MEMS microphone comprises an acoustic transducer for receiving an external ultrasonic signal and converting the ultrasonic signal into an electric signal; an ASIC (Application Specific Integrated Circuit) chip, coupled to the acoustic transducer, wherein the ASIC chip comprises: an amplifier unit for performing amplification on the electric signal and outputting an amplified signal; a decoding unit, connected to the amplifier unit, and configured to decode the amplified signal to obtain a decoded signal sequence; a matching unit, connected to the decoding unit, and configured to match the decoded signal sequence with a predetermined identification code to obtain a matched signal; a control unit, connected to the matching unit, and configured to generate one or more circuit adjusting parameters under the action of the matched signal.

Abstract: Microphone capsules for condenser or electret microphones often exhibit individual deviations from a desired ideal behavior, e.g. the frequency response and phase response. Particularly if a plurality of microphone capsules are interconnected to form a microphone array, suitable microphone capsules must be found in a selection process. Some of these deviations can be corrected electronically, e.g. by filtering with a corresponding filter that is individually adapted. An improved microphone capsule, with which an automatic selection and automatic assembly of circuit boards with microphone capsules is facilitated, comprises an electrostatic sound transducer, an amplifier element that outputs an amplified output signal of the electrostatic sound transducer, and at least one electronic memory element. Data obtained by a measurement and relating to the individual frequency response or phase response of the respective microphone capsule can be stored therein.

Abstract: A method and multimedia device for calibrating audio delays of an audio system with wireless loudspeakers are disclosed. The method allows to emulate surround or 3D sound systems with spatially distributed wireless loudspeakers. A first probe signal for audio playback, having a first audio pattern, is transmitted by a loudspeaker. A signal quality is determined by listening to an acoustic response of the loudspeaker to the first probe signal. A second audio pattern is determined by modifying the first audio pattern based on the signal quality. A second probe signal for audio playback having the second audio pattern is transmitted by the loudspeaker. An audio delay compensation is then determined for the loudspeaker based on a measure of an audio delay between the transmitted second probe signal and an acoustic response of the loudspeaker in response to the second probe signal.

Abstract: Techniques are disclosed for providing a virtual demonstration experience that includes presenting a user interface including an option to select between at least two configurations. The techniques also include initiating a first playback of an audio track for a first configuration, the first playback based on a binaural recording of a first audio system setup rendering the audio track. The techniques also include, in response to selection of a second configuration while the first playback of the audio track is occurring, switching to a second playback of the audio track, the second playback based on a binaural recording of a second audio system setup rendering the audio track, the second audio system setup different from the first audio system setup, wherein switching to the second playback of the audio track continues from where the first playback was prior to switching to the second playback.

Abstract: Systems and methods for detecting and configuring passive speakers within a playback system using a graphical user interface are disclosed. In one embodiment, a method of for detecting and configuration passive speakers in a playback system using a mobile device includes deriving speaker identification data concerning one or more passive speakers connected to an audio device in a playback system based upon at least an electrical signal sent to and returned from the one or more passive speakers, where the electrical signal is sent by the audio device including an audio stage comprising one or more amplifiers, and where the speaker identification data comprises information identifying a type of speaker, and displaying a graphical user interface screen on a mobile device based upon the identified type of speaker, where the displayed information and selectable options are dependent upon the identified type of speaker.

Abstract: A method for determining a voice coil position of a voice coil includes providing a magnetic circuit having a magnetic gap and suspending the voice coil in the magnetic gap, applying a driving signal to the voice coil to produce an electromotive force, providing inductive sensors mechanically coupled to the voice coil, measuring inductive sensor signals based on outputs from the inductive sensors, processing the measured inductive sensor signals by determining at least one inductive sensor signal ratio, and determining a representation of the voice coil position based on the at least one inductive sensor signal ratio. A voice coil system, which can be incorporated in a loudspeaker, is configured to carry out the method.

Abstract: A test and measurement instrument including a voltage source configured to output a source voltage, a current sensor, and one or more processors. The one or more processors are configured to determine an estimation of a load of an unknown connected device under test based on the source voltage, the current sensor, and a voltage of the connected device under test without any prior knowledge of the connected device under test.

Abstract: The disclosure provides a method and a host for adjusting audio of speakers, and a computer readable medium. The method includes: controlling a far-field speaker to play a first audio signal; controlling an audio receiver to receive the first audio signal from the far-field speaker and accordingly positioning a speaker location of the far-field speaker; establishing a first hearing transfer function related to the far-filed speaker based on the speaker location of the far-filed speaker; controlling the far-field speaker to play a second audio signal based on a second hearing transfer function; controlling the audio receiver to receive the second audio signal and accordingly estimating a first reference hearing volume; obtaining a second reference hearing volume corresponding to a first near-field speaker; and adjusting a first volume of the far-field speaker based on the first reference hearing volume and the second reference hearing volume.

Abstract: An apparatus and method for determining signals representative of events in the environment of a reactive transducer while being driven by a switching amplifier is disclosed. While the switching amplifier is in a zero voltage state, a signal capture circuit that is also in a zero voltage state is connected to the transducer for a relatively brief period of time during which a measurement is made of the residual current flow due to the inductance of the transducer. A prediction of the output signal is then subtracted from the signal measured across the transducer, reducing the overall range of the signal and increasing the relative size of the back-EMF signal compared to any remaining output signal. If desired, conventional echo cancellation can then be performed. The back-EMF signal can then be subjected to further processing by an analog-to-digital converter as known in the art.

Abstract: A method performed by a control device includes transmitting, to a first device, a signal for configuring an audio data output volume of the first device as a reference value, transmitting, to the first device, a test stream for calibration on an audio data output volume of devices consisting of a multi-channel surround system, and measuring an audio data output of the first device based on the reference value and the test stream. Further operations include transmitting, to second and third devices, a signal for configuring an audio data output volume of each of the second and third devices as a specific value, transmitting the test stream to the second and third devices, and measuring audio data outputs of each of the second and third devices which is outputted based on the specific value and the test stream.

Abstract: Example techniques may involve performing aspects of a spatial calibration. An example implementation may include detecting a trigger condition that initiates calibration of a media playback system including multiple audio drivers that form multiple sound axes, each sound axis corresponding to a respective channel of multi-channel audio content The implementation may also include causing the multiple audio drivers to emit calibration audio that is divided into constituent frames, the multiple sound axes emitting calibration audio during respective slots of each constituent frame. The implementation may further include recording the emitted calibration audio. The implementation may include causing delays for each sound axis of the multiple sound axes to be determined, the determined delay for each sound axis based on the slots of recorded calibration audio corresponding to the sound axes and causing the multiple sound axes to be calibrated.

Abstract: An example computing device is configured to perform functions including receiving calibration data corresponding respectively to a plurality of playback devices, where each playback device in the plurality of playback devices is located in a respective playback environment other than a first playback environment. The functions also include receiving playback device characteristic data respectively indicating at least one playback device characteristic for each playback device of the plurality of playback devices. The functions also include, based on at least the calibration data and the playback device characteristic data, determining updated playback device configuration information and transmitting data indicating the updated playback device configuration information to a given playback device located in the first environment.

Abstract: Systems and methods of driving plate loudspeakers with different parameters based on frequency region in a way similar to typical cone driver crossover networks are described. These systems and methods may be implemented using arrays of independently controlled drivers which allow a designer to emphasize or de-emphasize certain modes in certain frequency bands. Tuning the characteristics of the plate's motion can also affect the acoustical properties in a larger space rather than just at a single location. The systems and methods described herein can grant a designer a degree of control over the characteristics and performance of the plate.

Abstract: Techniques described herein may involve audio settings based on an environment. An example implementation may involve a playback device playing back first audio content and during playback of at least a portion of the first audio content, detecting, via the at least one microphone, an audio signal. At least a portion of the detected audio signal may include a reflection of the first audio content played back by the playback device. The playback device may determine an equalization setting based on at least the determined one or more reflection characteristics and apply the determined equalization setting during playback of second audio content.

Abstract: An active noise control device includes a control signal generating unit including a first adaptive filter configured to generate a control signal by performing a filtering process on a reference signal corresponding to noise, an identifying unit configured to identify a peak frequency in an impedance frequency characteristic of an actuator, a peak frequency storage unit configured to store an initial peak frequency of the actuator, a first determination unit configured to determine whether or not a difference between the peak frequency currently identified and the initial peak frequency is greater than or equal to a threshold value, and a control unit configured to change a characteristic of the control signal generated by the control signal generating unit when the first determination unit determines that the difference is greater than or equal to the threshold value.

Abstract: The disclosure relates in general to on-ear transition detection, and in particular to on-ear transition detection circuitry comprising: a monitoring unit operable to monitor a speaker current flowing through a speaker and/or a speaker voltage induced across the speaker, and to generate a monitor signal indicative of the speaker current and/or the speaker voltage; and an event detector operable to detect a qualifying disturbance in a sensor signal indicative of a qualifying pressure change incident on the speaker caused by the speaker transitioning from an on-ear state to an off-ear state or vice versa, wherein the sensor signal is, or is derived from, the monitor signal.

Abstract: An automated system includes transducers, at least one computing device, and at least one automated apparatus. The transducer(s) is/are driven and sensed using drive-sense circuit(s). A drives and senses drive and sense a transducer via a single line, generates a digital signal representative of a sensed analog feature to which the transducer is exposed, and transmits the digital signal to the computing device. The computing device receives digital signals from at least some of drive-sense circuits and process them in accordance with the automation process to produce an automated process command. The automated apparatus executes a portion of an automated process based on the automated process command.

Abstract: Methods and systems are provided for a portable speaker arranged in a vehicle. In one example, a system comprises a speaker configured to dock mechanically to a docking station in an area of a vehicle, wherein the speaker further comprises a removable battery.

Abstract: Monitoring circuitry, comprising: a current monitoring unit operable to monitor a speaker current flowing through a speaker and generate a monitor signal indicative of that current; and a controller operable, based on a control signal, to control the current monitoring unit to control whether the monitor signal is generated and/or a property of the monitor signal.

Abstract: An auditory device assembly (1) comprises an earpiece (3) having an audio output device (7) for an ear (9) of a user and an audio processing unit (11, 33). The audio processing unit (11, 33) has a hearing-test mode and an audio streaming mode. In the hearing test mode the audio processing unit (11, 33) is arranged to determine at least one ear characteristic of the ear of the user based on at least one hearing test. In the audio streaming mode the audio processing unit (11, 33) is arranged to output an audio stream via the audio output device (7). The audio processing unit (11, 33) is arranged to adjust the output of the audio stream in the audio streaming mode based on the at least one ear characteristic determined in the hearing-test mode.

Abstract: A phase optimizer optimizes, for each listening position in a listening environment, a phase shift for each frequency in a range of predetermined frequencies. The phase optimizer determines a resultant phase value for each possible phase shift value and stores the resultant phase values for each possible phase shift value in an array. The phase optimizer calculates mean and standard deviation for the resultant phases stored in the array. The mean and standard deviations stored in the array are compared and phase shift values that result in the resultant phase values having the smallest mean and standard deviations are selected and are stored in memory. The phase optimizer optimizes each frequency, within a predetermined range of frequencies, for all possible phase shift values within a predetermined range of phase shift values and generates a phase shift target curve generated to be output by the phase optimizer.

Abstract: Systems and methods are provided for using one or more external Bluetooth speakers in conjunction with an in-vehicle Bluetooth-enabled audio system. Locational features or aspects of the external Bluetooth speakers and the in-vehicle Bluetooth-enabled audio system can be leveraged in a manner that allows for automatic or manual adjustment of the one or more external Bluetooth speakers, e.g., adjusting left/right balance of the one or more external Bluetooth speakers based on the position or location of the or more external speakers relative to the vehicle, in-vehicle Bluetooth-enabled audio system, etc.

Abstract: An internet protocol (IP)-based ceiling or wall-mounted speaker with optional IP-based camera and/or alarm indicator is provided. The IP-based speaker has a detachable add-on device for accommodating different types of interchangeable camera configurations and other components such as flush mount camera, camera providing angled view, night vision-type camera for different services and configurations. Multiple IP-based speakers are connected to an IP device to exchange audio data via an Ethernet connection for cost effective, flexible and convenient installations. The IP-based speaker has a speaker cone for talkback features, and relay for remote relay control of doors or gates for security applications as well as public address applications.

Abstract: One embodiment provides a computer-implemented method that includes receiving a trigger sound from a primary listening location. The trigger sound being received at multiple speakers in a synchronous network at different times. The trigger sound is recognized at the multiple speakers. A respective relative delay is determined based on a time differential function that determines time differences. Sound quality for the multiple speakers is improved based on the respective relative delay for each of the multiple speakers.

Abstract: Embodiments described herein generally relate to measuring and evaluating a test signal generated by a device under test (DUT). In particular, the test signal generated by the DUT may be compared to a reference signal and scored based on the comparison. For example, a method may include: capturing a test signal from a device under test; splicing the test signal into a plurality of test audio files based on a plurality of frequency bins; at each frequency bin, comparing each of the plurality of test audio files to a corresponding reference audio file from among a plurality of reference audio files, the plurality of reference audio files being associated with a reference signal; and calculating a performance score of the device under test based on the comparisons.

Abstract: A recurrent neural network is employed in a loudspeaker system to compensate the distortion of the system based upon a source signal (content) and the sensing output of a sensing circuit (context). A frequency domain transform is selected to provide mapping between the source signal and a recorded signal; and enable reconstruction of desirable playback. Various sensing-related features and source-related features are derived to serve as the auxiliary information. A desirable content is therefore generated based upon the original content and the context.

Abstract: Embodiments of systems and methods are described for estimating a position of a loudspeaker and notifying a listener if an abnormal condition is detected, such as an incorrect loudspeaker orientation or an obstruction in a path between the loudspeaker and a microphone array. For example, a front component of a multi-channel surround sound system may include the microphone array and a position estimation engine. The position estimation engine may estimate the distance between the loudspeaker and the microphone array. In addition, the position estimation engine may estimate an angle of the loudspeaker using a first technique. The position estimation engine may also estimate an angle of the loudspeaker using a second technique. The two angles can be processed to determine whether the abnormal condition exists. If the abnormal condition exists, a listener can be notified and be provided with suggestions for resolving the issue in a graphical user interface.

Abstract: Systems and methods for determining sound-producing characteristics of electroacoustic transducers are disclosed. According to an aspect, a system includes electroacoustic transducers configured to generate sound. The system also includes an acoustoelectric transducer configured to convert sound produced by the electroacoustic transducers into one or more electrical signals. Further, the system includes a computing device configured to apply one or more patterns of electrical signals to the electroacoustic transducers to test for one or more sound-producing characteristics. The computing device is also configured to receive, from the acoustoelectric transducer, electrical signals that resulted from application of the patterns of electrical signals to the electroacoustic transducers.

Abstract: Embodiments are disclosed for disabling/re-enabling head tracking for spatial audio applications. In an embodiment, a method comprises: obtaining, using one or more processors of an auxiliary device worn by a user, motion data; tracking, using the one or more processors, the user's head based at least in part on the motion data; determining, using the one or more processors, whether or not the user is walking based at least in part on the motion data; in accordance with determining that the user is walking, determining if a source device configured to deliver spatial audio to the auxiliary device is static for a specified period of time; and in accordance with determining that the user is walking and the source device is static for the specified period of time, disabling the head tracking.

Abstract: A loudspeaker controller for estimating a fundamental resonance frequency of a loudspeaker includes: an amplifier circuit, arranged to generate a driving signal of the loudspeaker according to an audio input signal; a sensing circuit, arranged to sense characteristics of the driving signal to generate a measurement signal; a plurality of band pass filter circuits, arranged to filter the measurement signal to generate a plurality of filter outputs, respectively, wherein the plurality of band pass filter circuits have different passbands; and an estimation circuit, arranged to estimate the fundamental resonance frequency according to the plurality of filter outputs.

Abstract: An example method of operation includes identifying a loudspeaker array profile defining characteristics of a loudspeaker array stored in memory, identifying a three-dimensional venue geometry value stored in the memory, defining virtual receivers to simulate acoustic characteristics within the venue geometry, defining a number of passive acoustic filter permutations to perform within a range of passive acoustic filter settings, and each passive acoustic filter setting is unique and has one or more passive acoustic filters to apply to one or more loudspeakers in the loudspeaker array, selecting performance criteria to apply to the loudspeaker array to represent its sound coverage uniformity at a given location throughout the venue geometry, calculating the performance criteria of the loudspeaker array via a passive acoustic filter setting selected from one or more of the passive acoustic filter permutations by performing a simulation with the passive acoustic filter settings, identifying an optimized passive

Abstract: A calibration device and method of manufacturing the same. The calibration device for a hearing earpiece configured to at least partially protrude into the ear canal of a user includes a calibration base. The calibration base a calibration insert. An air chamber is defined between the calibration base and the calibration insert. The calibration device also includes at least one earpiece receiving mechanism defined on the calibration insert. The at least one earpiece receiving mechanism is configured to create an sealed connection between given ear piece and the air chamber for calibration. The earpiece receiving mechanism is configured to at least partially receive the given earpiece. A corresponding method of manufacturing is also included.

Abstract: A two-channel balance method and an electronic device using the same are provided. The two-channel balance method includes the following steps. A gain-frequency information of a two-channel signal is adjusted. A sampling delay information of the two-channel signal is calculated according to a distance information among a sound receiving unit, a left speaker unit and a right speaker unit. A forward test audio file or a surround test audio file is generated according to the sampling delay information. A phase offset information is estimated according to at least the forward test audio file or the surround test audio file. A phase offset direction information is determined. A phase information of the two-channel signal is adjusted according to the phase offset information and the phase offset direction information.

Abstract: Disclosed is a device for processing an audio signal, which renders an audio signal. The device for processing an audio signal includes a processor. The processor receives metadata including an audio signal and first element reference distance information and renders a first element signal on the basis of the first element reference distance information, wherein the first element reference distance information indicates the reference distance of an element signal. The audio signal is capable of including a second element signal which may be simultaneously rendered with the first element signal, and the metadata is capable of including second element distance information indicating the distance of the second element signal. The number of bits required for representing the first element reference distance information is smaller than the number of bits required for representing the second element distance information.

Abstract: A sound collection apparatus includes a substantially spherical base on which at least a predetermined number of recess portions are provided in a surface thereof with a predetermined interval therebetween; and a predetermined number of microphones. The predetermined number is an integer of 2 or greater. The microphones are installed on an inner bottom surface side of the recess portions one by one. Intervals between sound collecting portions of adjacent microphones are substantially equal to each other.

Abstract: A method of processing a radio frequency signal includes: receiving the radio frequency signal at an antenna of a receiver device; processing, by a radio frequency front-end device, the radio frequency signal; converting, by an analog-to-digital converter, the analog signal to a digital signal; receiving, by a neural network, the digital signal; and processing, by the neural network, the digital signal to produce an output.

Abstract: A method for delivering an optimized acoustic signal to a targeted sound field includes determining a bright zone having a high acoustic energy sound delivery region and a dark zone having low acoustic energy sound delivery region. The system selects a finite impulse response (FIR) filter set associated with a plurality of loudspeakers, and generates, via the plurality of loudspeakers and based on the FIR filter set, an acoustic signal having a sound pressure due to the combined loudspeakers (p) at an observation point r in a sound field disposed in the bright zone. The acoustic signal is delivered such that the sound pressure due to the combined loudspeakers at the observation point is approximately equal to unity acoustic energy at the observation point when the sound pressure due to the combined loudspeakers is generated via a single loudspeaker of the plurality of loudspeakers, and the remaining loudspeakers of the plurality of loudspeakers are off.

Abstract: Provided are a sounding structure and a terminal. The sounding structure comprises: a frame comprising an accommodation cavity, the accommodation cavity comprising a first opening and a second opening provided on two opposite sides of the frame; a magnetic circuit system arranged in the accommodation cavity and comprising a first channel and a second channel; a first vibrating diaphragm arranged to face the first opening and provided with a voice coil, the voice coil partially extending into the first channel; and a second vibrating diaphragm arranged to face the second opening, the second channel communicating the first opening with the second opening.

Abstract: A speaker drive unit includes a control signal generation unit and a signal processing unit. The control signal generation unit generates a control signal on the basis of detection signals generated by a plurality of detection units that is disposed in one-to-one correspondence with a plurality of actuators that vibrates a diaphragm, and detects vibration conditions. The signal processing unit includes a plurality of signal systems corresponding one-to-one to the plurality of actuators. The signal systems provide a drive signal to the corresponding actuators on the basis of the same audio signal. The drive signal generated by at least one signal system of the signal systems is controlled by the control signal.