Harman International Industries Patente

An on-wall speaker system includes a body, a first pair of transducers, and a high frequency actuator element. The body includes a plurality of side walls and a front wall. The first pair of transducers are supported by first opposing side walls of the plurality of side walls. The high frequency actuator element is mounted to the front wall of the body and is configured to induce mechanical vibrations in the front wall to generate high frequency sound waves to a listening environment.

The present disclosure provides a light source control method, a light-emitting assembly, an audio device, and a computer-readable storage medium. The light source control method includes: acquiring light-emitting pattern data for controlling light-emitting characteristics of a light source array, the light source array including a plurality of light sources; measuring a temperature of the light source array in real time using a temperature sensor; determining, based on the light-emitting pattern data and the temperature of the light source array, to adjust the light source array from a first average brightness level to a second average brightness level; and controlling light-emitting characteristics of each light source in the light source array non-proportionally based on the second average brightness level and the light-emitting pattern data, such that the light source array displays a light-emitting pattern corresponding to the light-emitting pattern data while satisfying the second average brightness level.

A loudspeaker includes a speaker enclosure, first and second speaker drivers housed within the speaker enclosure, and first and second acoustic transmission lines housed within the speaker enclosure. The first and second acoustic transmission lines are respectively associated with the first and second speaker drivers and are a conjoined pair of acoustic transmission lines. The loudspeaker may be arranged within a structure of an assembly. The assembly may be a vehicle seat with the structure being a head rest or an upper portion of the vehicle seat.

A shield assembly arranged around an acoustic sensor may include an outer shell extending from an outer apex to a base and forming a dome-shape, the outer shell defining at least one perforation to allow air to flow therethrough, and an inner shell extending from an inner apex to the base inside of the outer shell, the inner shell creating a channel between the outer shell, inner shell, and base to receive airflow at the at least one perforation, wherein the at least one perforation and channel redirect air to protect a microphone arranged within the inner shell from external airflow and preserving sound pressure measurement integrity.

A data processing method and a control device are provided. The method of the present disclosure is directed to a loudspeaker system including a plurality of loudspeakers. By collecting locally measured distance information from respective loudspeakers and determining a real-time spatial layout of the loudspeaker system based on the distance information, automatic layout calibration of the loudspeaker system is achieved. Subsequently, according to the layout of the loudspeaker system, and in combination with an actual orientation of a user and spatial position-related data indicative of a position of a sound source in audio content to be played, corresponding data in the audio content is assigned to corresponding loudspeakers, thereby utilizing playback by the respective loudspeakers to collectively achieve spatial audio rendering, providing an immersive sound experience for the user. The method of the present disclosure does not require additional layout calibration processing, and thus can achieve real-time layout determination without user awareness and can automatically adjust audio playback of the loudspeakers according to changes in the real-time orientation of the user, thereby achieving user orientation-adaptive spatial audio rendering.

Various embodiments disclose techniques for generating one or more sound zones and a quiet zone in a listening environment. The techniques include configuring one or more filters of a filter bank, generating a respective speaker signal for each of one or more speakers by processing a respective audio zone signal for each of one or more sound zones using the filter bank, and outputting the respective speaker signals using the one or more speakers. Sound output by the one or more speakers propagates through a listening environment to generate respective sound in each of the one or more sound zones corresponding to the respective audio zone signal for that sound zone. The sound output by the one or more speakers propagates through the listening environment to generate a first quiet zone.

The present invention provides a microphone module, comprising: a housing; a microphone unit disposed in the housing and located at the bottom of the housing; a top sound hole area located at the top of the housing; a first windproof net covering the top sound hole area; a windproof cotton unit disposed between the top sound hole area 5 and the microphone unit; and a second windproof net disposed between the windproof cotton unit and the microphone unit. The present invention also provides a wearable device, including the microphone module as described above. This invention provides a microphone module with excellent windproof and noise-reducing functions, which can ensure clear call quality in windy weather with wind speeds of up to 9 m/s or even 11 m/s. In addition, the microphone module is small and modular, and can be embedded in wearable devices of any shape.

The present disclosure relates to a loudspeaker assembly (100), characterized in that the loudspeaker assembly (100) comprises: a magnetic circuit unit (30) comprising a top plate (36) and defining a magnetic gap (38); a frame unit (20) connected to the magnetic circuit unit (30); and a vibration unit (10) connected to the frame unit (20) and comprising an upper diaphragm (12), a voice coil (14), and a dome (16), the voice coil (14) extending at least partially into the magnetic gap (38), wherein the dome (16) comprises a dome main body (160) and at least two crimped edge portions (162, 164) surrounding the dome main body (160), wherein the dome (16) is affixed to a top side of the voice coil (14) by means of the crimped edge portions (162, 164), such that the top plate (36) of the magnetic circuit unit (30) is disposed in a middle of the voice coil (14) in a height direction (z).

In at least one embodiment, a feedback amplifier is provided. The amplifier includes an input stage circuit (206) and a collector element (208). The input stage circuit (206) is configured to receive at least one input feedback signal having wide band frequency components that include high frequency components and low frequency components. The input stage circuit (206) includes a first gain stage (220), a filter, and a second gain stage (222). The first gain stage (220) amplifies the at least one input feedback signal to generate at least one amplified output signal. The filter filters the wide band frequency components on the at least one input feedback signal to generate at least one filtered feedback signal having a first low frequency component. The second gain stage (222) amplifies the at least one filtered feedback signal having the first low frequency component and to provide at least one gain output signal.

Provided are a device for generating a star-sparkling lighting effect and a method thereof. The method includes arranging at least one multi-color light source (110) to emit multiple lights (112, 114, 116) of at least two colors, and arranging at least one light-guiding lens (130) in front of the multi-color light source, such that the multiple lights are mixed into mixed-color lights in a diffuse lens portion (150) of the light-guiding lens, and then the mixed-color lights are divergently emitted to a front grille panel (140) via a reflective lens portion (160), and through gaps in the front grille panel, a star-sparkling lighting effect is generated.

Provided are an audio peak limiting system and method, which, by setting a moving maximum window to drive gain calculation, and in combination with matching a moving maximum window length to a delay duration of an audio input signal, suppress a risk of overshoot while enabling a user to flexibly adjust a suitable attack time, and result in a smoother gain curve and reduce distortion in dynamic range processing, thereby achieving a more natural auditory perception.

Methods and systems for configuring an acoustic device network are provided. In one example, a method comprises: connecting a first device (e.g., a mobile device with audio control software) and a second device (e.g., a primary audio device) to a network; transmitting a first audio signal with a third device (e.g., a secondary audio device), wherein the first audio signal is encoded with a network address of the third device. This first audio signal may be detected and the network address decoded on the second device; the second device may transmit a connection request to the first device; in response to a confirmation of connection received by the first device, a second audio signal may be transmitted with the second device, wherein the second audio signal is encoded with network credentials of the network; this second audio signal may be detected and the network credentials decoded with the third device.

A data processing method, apparatus, and device, a computer program product, and a computerreadable storage medium are provided. With the method, microphone data is shared between a plurality of speakers by using a data connection between the plurality of speakers, to determine an awakened AI speaker of the plurality of speakers based on the shared microphone data, then AI audio data is generated for the captured microphone data by using an AI functional module in the awakened AI speaker, and the generated AI audio data is shared to at least part of the plurality of speakers, thereby implementing sharing of an AI speaker functionality between a plurality of speaker devices. By implementing sharing of an AI speaker functionality between the plurality of speaker devices, each speaker is enabled to support more AI functionalities than its original configuration, and even a speaker configured with no any AI functionality can still support all AI functionalities of all of the plurality of speakers.

A compression driver for an omnidirectional loudspeaker comprises a magnet assembly disposed about a central axis, an annular diaphragm disposed coaxially above and operably connected to the magnet assembly, a phasing plug including a base portion having a first side and an opposed second side facing the diaphragm, the base portion including a plurality of apertures that extend therethrough and are arranged generally circumferentially about the central axis, the phasing plug including a raised portion extending upwardly from the base portion and defining a plurality of radially-expanding channels acoustically connected to the apertures, and a compression chamber defined between the diaphragm and the phasing plug, wherein actuation of the diaphragm by the magnet assembly generates sound waves within the compression chamber which travel through the plurality of apertures and the radially-expanding channels to create a generally horizontal 360° radiation pattern of the sound waves from the compression driver.

A phasing plug for a compression driver includes a body having an inlet side with a front surface and an outlet side with a rear surface, the body disposed about a central axis. A plurality of channels are formed through the body from the inlet side to the outlet side, each of the plurality of channels having an annular configuration with an entrance at the front surface, an exit at the rear surface, and a path length between the entrance and the exit. The plurality of channels have unequal path lengths such that acoustical signals traveling through the plurality of channels form a convex wavefront at the outlet side of the phasing plug.

A robust range prediction system is proposed that estimates the range of an electric vehicle (EV) during operation, using a range prediction model trained on data relevant to the energy usage of the vehicle. The range prediction model may take a greater number of inputs than may be considered by an OEM range estimator provided with the EV. The predicted range generated by the range prediction system may then be compared with the estimated range of the OEM range estimator, and/or combined into an integrated display to provide the driver with a confirmation of the estimated range, which may decrease a range anxiety of the driver. Additionally, the proposed range prediction system may generate recommendations for how to maximize (e.g., maintain or increase) the predicted range by adjusting the operation of the vehicle or coordinating operation of the vehicle with other vehicles, traffic signals, etc.

Techniques are described for communications for audio systems. Some techniques involve receiving, by a protocol layer, a command from a first application layer of the first computing device, mapping the command to a protocol code and a parameter, and transmitting data that includes the protocol code and the parameter through an audio network, for example, to a second computing device.

A computing device includes a display; and at least one processor programmed to transmit a multimedia signal including one or more of audio and video data for output via one or more of a loudspeaker and the display; receive a first user input via on a first interface of the display, the first user input being indicative of a user selection to create personalized content including a start position and an end position corresponding to a portion of the multimedia signal; present at least one attribute of the personalized content on a second interface for selection by a user; and associate the at least one attribute to the personalized content of the multimedia signal to provide a desired attribute in response to a second user input being received at the second interface, the second user input being indicative of selecting at the at least one attribute.

An in-car audio system, may include at least one vehicle speaker arranged in a vehicle headrest behind an occupant and configured to transmit audio signals to the associated occupant, the at least one vehicle speaker including a pair of speakers configured to provide binaural audio signals to the associated occupant; at least one visual indicator arranged within a vehicle cabin at a first location and configured to provide a visual cue to the occupant; a processor programmed to instruct the speaker to emit an audio signal perceived at the first location, and instruct the at least one visual indicator to concurrently illuminate with the audio signal to provide visual reinforcement to the occupant that the audio signal perceived is associated with the first location.

An audio system is provided. The system includes a mobile device and a hub controller device. The mobile device transmits calibration data. The hub controller device receives an audio input signal indicative of desired audio to be played back in an establishment and processes the audio input signal based on the calibration data to generate an audio output signal for playback in the establishment. The hub controller device transmits the audio output signal to at least one of a wireless loudspeaker and wireless headphones to playback the audio output signal in the establishment and transmits the audio output signal, via a wired connection, to a hardwired based loudspeaker device to playback the audio output signal in the establishment.

Systems and methods are herein provided for dynamic integration of portable audio devices into a vehicle audio system. In one example, a vehicle audio system comprises one or more processors; and memory storing instructions that when executed cause the one or more processors to: recognize one or more portable audio devices paired to the vehicle audio system via a companion application; establish a communication link to each of the one or more portable audio devices; determine an audio output configuration for each of the one or more portable audio devices; integrate each of the one or more portable audio devices into the vehicle audio system; and output audio content via the one or more portable audio devices and one or more cabin output speakers of the vehicle audio system according to the audio output configuration for each of the one or more portable audio devices.

In at least one embodiment, a system for monitoring driver behavior is provided. The system includes a plurality of sensors and at least one controller. The plurality of sensors transmits a first signal indicative of whether a first driver is in a distracted state. The at least one controller is programmed to receive a first driver identity signal that identifies the first driver before receiving the first signal and to receive a first mitigation signal indicative of a first selected mitigation operation from a plurality of mitigation operations for the first driver. The at least one controller is further programmed to perform the first selected mitigation operation for the first driver in response to at least the first signal indicating that the first driver is in the distracted state.

The present disclosure relates to a loudspeaker assembly. The loudspeaker assembly (100) includes: a backplate unit (24), the backplate unit (24) including a backplate (240) and a snap hook (244) arranged in the backplate (240); and a hanging device (26), the hanging device (26) including a hanging portion (260) and a snap-fit member (264), where the backplate (240) includes a snap-fit member guide opening (2404), the snap-fit member (264) being insertable into the snap-fit member guide opening (2404), and where the hanging device (26) is switchable between a snap-locked position and a released position, where in the snap-locked position, the snap-fit member (264) is maintained in snap-fit connection with the snap hook (244), thereby preventing separation of the hanging device (26) from the backplate unit (24); and in the released position, the snap-fit connection of the snap-fit member (264) with the snap-hook (244) is released, thereby allowing separation of the hanging device (26) from the backplate unit (24).

In one embodiment, a computer-implemented method includes receiving an input audio signal for playback by at least one speaker and at least one tactile transducer, receiving a control signal associated with a user-specified setting for the at least one tactile transducer, and generating a processed audio signal to drive the at least one tactile transducer, wherein the processed audio signal is adjusted based on the control signal independent of a speaker output signal driving the at least one speaker.

Methods and mechanisms are provided for emulating automotive audio bus (A2B) based accelerometers. A port of an electronic device, which may be implemented using a Field-Programmable Gate Array (FPGA), may be connected downstream from an A2B port of an A2B node of a vehicle. A set of emulated accelerometer data at the electronic device may be selected, and the set of emulated accelerometer data may then be transmitted from the A2B port of the electronic device to the A2B port of the A2B node.

The present disclosure relates to a speaker box assembly, where the speaker box assembly includes: a speaker box housing, the speaker box housing being cylindrically constructed and including a circumferentially surrounding housing wall and an end opening defined by the housing wall; at least one diaphragm assembly, the diaphragm assembly including a diaphragm unit and a diaphragm surround surrounding the diaphragm unit; and at least one basket device, the diaphragm assembly being supported in the speaker box housing by means of the basket device, where the basket device includes a conical section and an end section, where an outer edge of the end section extends to an inner wall of the housing wall, thereby causing the basket device to be arranged in the end opening of the speaker box housing in a form-fitting manner by means of the end section, where the end section constitutes a heat dissipation surface for releasing heat generated during operation of the speaker box assembly to an environment.

The present inventive subject matter provides a method for tuning ANC based on a hearing threshold level of a user and an apparatus for tuning ANC based on a hearing threshold level of a user, wherein hearing threshold level data of the user is obtained by performing a hearing test on the user, and an ANC filter is tuned based on the hearing threshold level data of the user, so that the user obtains more optimized ANC effects.

A speaker includes a magnetic assembly having top and bottom air gaps that are spaced apart from one another along an axis. The speaker further includes a voice coil assembly having a bobbin with top and bottom voice coils. The bobbin is movable in upwards and downwards directions along the axis relative to the magnetic assembly with at least one of the voice coils being at least partially positioned respectively within at least one of the air gaps. The voice coils are spaced apart from one another along the axis with an outward offset such that in a mean axial position of the bobbin a larger portion of the top voice coil extends upwards above the top air gap than downwards below the top air gap and a larger portion of the bottom voice coil extends downwards below the bottom air gap than upwards above the bottom air gap.

Provided are a method for setting a sequence number of a light-emitting device, a light-emitting device, and a medium. The method may be implemented by a first light-emitting device, wherein the first light-emitting device establishes a short-range communication connection with a computing device, and the method comprises: obtaining a device setup command from the computing device via the short-range communication connection, and entering a device setup mode in response to the device setup command; broadcasting a notification in response to the device setup command to notify remaining light-emitting devices other than the first light-emitting device in a predetermined environment to enter the device setup mode; setting, in response to a user input, a first sequence number of the first light-emitting device based on whether sequence number information regarding sequence numbers is received from the remaining light-emitting devices; and broadcasting first sequence number information regarding the first sequence number.

A magnetic assembly for a speaker includes a first pole piece having a first pole tip and a second pole piece having a second pole tip. The pole pieces are spaced apart with the pole tips being opposed across an air gap. Each pole tip is flared toward the air gap. The magnetic assembly may further include a first permanent magnet in contact with the first pole piece and a second permanent magnet in contact with the second pole piece. The magnetic assembly may be a part of an electromagnet transducer further having a voice coil positioned at least partially within the air gap.

The present disclosure relates to a magnet assembly (144) for an audio speaker (100). The magnet assembly (144) comprises a magnet arrangement (132) including a through hole (142) and a light guide (104). The light guide (104) is configured to guide light through the through hole (142) of the magnet arrangement (132). The light guide (104) includes a first portion (202) having a cylindrical shape with a longitudinal axis (302). The first portion (202) has a light receiving end surface (204) for receiving light from a light source (106) and is configured to be inserted in the direction of the longitudinal axis (302) into the through hole (142) of the magnet arrangement (132). The light guide (104) includes a second portion (206) for emitting the light received at the light receiving end surface (204). The light guide (104) includes a first protrusion (216) protruding perpendicular to the longitudinal axis (302) from the lateral surface of the first portion (202). The first protrusion (216) serves as a support structure abutting against an edge of the through hole (142) when the light guide (104) is inserted into the through hole (142). The light guide (104) includes at least one second protrusion (218, 220, 222) protruding perpendicular to the longitudinal axis (302) from the lateral surface of the first portion (202). The at least one second protrusion (218, 220, 222) is arranged, in the direction of the longitudinal axis (302), between the first protrusion (216) and the light receiving end surface (204). The at least one second protrusion (218, 220, 222) is dimensioned to provide an interference fit of the first portion (202) within the through hole (142) of the magnet arrangement (144).

An apparatus for displaying a volume level in real time while a microphone picks up an audio signal is provided. The apparatus includes a first stage amplifier amplifying an audio input signal into a first amplified signal, and a second stage amplifier amplifying the first amplified signal into a second amplified signal. The apparatus further includes a volume detection module comparing the first amplified signal and the second amplified signal with a volume display threshold to determine a volume level to be displayed, and a volume display unit displays the volume level in real time. A method for displaying a volume level in real time while a microphone picks up an audio signal is also provided.

An audio management method, a computing device, an audio system, and a medium are provided. The audio management method is performed by a computing device, and may include: sending, via communicative connection with one or more audio devices in an environment, an activation instruction to the one or more audio devices to activate a microphone included at the one or more audio devices to capture audio data; receiving, via the communicative connection, the captured audio data from the one or more audio devices, and performing speech recognition on the received audio data; and generating prompt control information in response to a target sound being recognized from the audio data based on the speech recognition.

A power supply and a method of operating a power supply. The power supply including a power input operable to receive a mains electric power, computing means comprising a processor, a first power supply unit (PSU) electrically coupled to the computing means, the first PSU operable to receive the mains electric power when the power supply is switched on and to provide power to the computing means. The power supply including a second PSU, a first switching means electrically coupled to the power input, to the first PSU and to the second PSU, the first switching means operable to divert the mains electric power to the first PSU or to the second PSU. The power supply including a second switching means operable to select a high voltage setting or a low voltage setting of the second PSU. The computing means are further electrically coupled to the first switching means and to the second switching means, and the processor is operable to measure the voltage of the mains electric power input to the first PSU. The processor is further operable to determine from the voltage measurement that the mains electric power is in a low voltage range or in a high voltage, wherein the low voltage range does not overlap with the high voltage range. The processor is further operable to operate the second switching means to the low voltage setting if the mains electric power is in the low voltage range or to the high voltage setting if the mains electric power is in the high voltage range. The processor is further operable to operate the first switching means, subsequent to operating the second switching means, from the first PSU to the second PSU to divert the mains electric power from the first PSU to the second PSU.

A method and electronic consumer device for creating an individual spatial listening zone for a user provided. The method includes receiving an input audio signal and processing it to generate first and second ultrasonic electric audio signals. The ultrasonic signals are provided to first and second ultrasonic transducers which convert them into directional ultrasonic acoustic signals. These signals are directed to a spatial listening zone where the user is located. Through non-linear interaction of the ultrasonic signals in air within the spatial listening zone, an audible sound signal is produced for the user that corresponds to the desired audio based on the input signal. The audible sound is substantially confined to the spatial listening zone and inaudible outside it. Various techniques for dynamically adjusting the spatial listening zone based on user position are disclosed. The system allows for creation of personalized audio experiences while minimizing disturbance to others nearby.

A system for a vehicle includes a wireless transceiver configured to communicate with a mobile device via a wireless connection encrypted using a digital key, wherein the digital key allows the mobile device to access a driving function of a vehicle; and one or more controllers programmed to responsive to a user input indicative of an intent to make a purchase by a user of the vehicle, transmit a request to the mobile device to launch a payment for the purchase via the wireless connection, perform an authentication by interacting with the user to authorize the payment, and responsive to the authentication, instruct the mobile device to proceed with the payment via the wireless connection.

Zusammenfassung wird geladen …

Zusammenfassung wird geladen …

Zusammenfassung wird geladen …

Zusammenfassung wird geladen …