11228 results about "Loudspeaker" patented technology

The invention discloses an intelligent control method and system for tunnel loudspeakers, and relates to the technical field of tunnel audio control, environmental parameters in a tunnel are collected by adopting a mode of deploying sampling equipment in a distributed manner, and data preprocessing is performed in a targeted manner for different environmental parameters; a sound propagation model is established, and attenuation and delay of sound in different environments are simulated. According to the intelligent control method and system for the tunnel loudspeakers, various temperature and humidity sensors are arranged in the tunnel, and the absolute humidity is calculated in combination with the air pressure data, so that the sound velocity is accurately corrected, and the phase difference of the multiple loudspeakers is reduced; an adaptive Kalman filtering algorithm is adopted to process wind speed data, and reliable input is provided for a sound propagation model; a deep reinforcement learning algorithm is used to carry out collaborative optimization on parameters such as amplitudes and directional angles of multiple loudspeakers, a Bayesian network is used to detect loudspeaker faults, and Delaunay triangulation and a distributed consistency algorithm are combined to realize rapid fault reconstruction.

A speaker apparatus including a housing; a slot loading loudspeaker inside the housing, the slot loading loudspeaker including a speaker unit, and a cover covering the speaker unit, and including an opening through which an acoustic signal from the speaker unit may be output; and a panel member disposed in a forward direction of the slot loading loudspeaker so that a gap is formed between the cover and the panel member, and an exposure area is formed between an edge of the housing and an edge of the panel member to expose the opening through the exposure area so that an acoustic signal outputted through the opening is transferred to an outside, wherein at least one of the cover or the panel member includes at least one blocking member for preventing at least a portion of the acoustic signal outputted through the opening from flowing into the gap.

A loudspeaker apparatus includes a circuit housing configured to accommodate a circuit component or a battery; an ear hook; a housing of an earphone core configured to accommodate the earphone core; and a housing protector at least partially covering a periphery of the circuit housing and the ear hook. A first end of the ear hook is connected to the circuit housing. The earphone core is driven by the circuit component or the battery to vibrate to generate sound. The housing of the earphone core is connected to a second end of the ear hook away from the circuit housing through a hinge component. The hinge component is capable of rotating to change a position of the housing of the earphone core relative to the ear hook.

The present disclosure relates to a speaker device. The speaker device may include an ear hook, a core housing for accommodating an earphone core, and a circuit housing for accommodating a control circuit or a battery. The ear hook may include a first plug end and a second plug end. The ear hook may be surrounded by a protective sleeve which may be made of an elastic waterproof material. The ear hook may be elastic, and a position of the core housing relative to the ear hook may be changed according to an elastic deformation of the ear hook, thereby the core housing may fit a user in front of or behind an ear of the user. The circuit housing may be fixed to the second plug end. The control circuit or the battery may drive the earphone core to vibrate to generate a sound.

A system may include a storage device, configured to store computer-executable instructions. A system may include an ear-bud configured to be positioned within an ear canal of a user, the ear-bud comprising: a speaker, a microphone; and one or more processors in communication with the storage device, wherein the computer-executable instructions, when executed by the one or more processors, cause the one or more processors to: obtain a user hearing profile, obtain an equal-loudness hearing profile, receive audio data from the microphone, and generate a second audio data based on a first sound-pressure level, a second sound-pressure level, a first frequency; and cause the speaker to emit the second audio data within the ear canal of the user, such that the user perceives the audio data as if the user has normal hearing.

An ear-worn device is provided that operates to isolate and individually treat the received speech of a target speaker or multiple target speakers from an audio input signal detected in a multi-speaker environment. The ear-worn device uses a machine learning model that receives a voice signature of each of one or more target speakers as input signals, to identify and isolate the component of the audio input signal attributable to the target speaker(s). Once isolated, the target speaker's speech may be enhanced, de-emphasized, or otherwise processed in a manner desired by the wearer of the ear-worn device. The wearer may use an external electronic device, e.g., a phone, to select one or more target speakers in a conversation and / or configure various settings associated with processing the speech on the ear-worn device.

To provide a game machine capable of advancing a game smoothly.SOLUTION: A game machine includes a main game unit that controls the progress of a game, a performance display unit that can display a performance, an operation member that can be operated by a player, a speaker that can output sound, and a sub-game unit that can be connected to earphones that can output sound. Connection to the earphones can be performed by operating the operation member by the player, and connection to the earphones can be disconnected by operating the operation member by the player. Connection to the earphones cannot be executed during symbol variation, but disconnection to the earphones can be executed during symbol variation.SELECTED DRAWING: Figure 802

Examples described herein relate to transitioning a playback session between portable playback devices such as “smart” headphones, earbuds, and handheld speakers with playback devices of a zone-based media playback system. Exemplary techniques facilitate continuity of playback when transitioning between locations (e.g., from at home to on-the-go or vice versa) or between listening paradigms (e.g., personal or out-loud playback of audio content). An example implementation includes detecting a swap trigger, determining the source playback device(s) and target playback device(s), and performing a playback session swap between the source playback device(s) and target playback device(s).

A system for serving personalized advertisements based on biometric and identifying data captured by sensors. The system employs pre-processing, feature extraction, and recognition modules to identify individuals. A context-aware processing module adapts recognition strategies based on environmental data and analyzes personal information to confirm the identity of the targeted individual and send targeted advertisements. The system includes display and speaker modules for visual and audio advertisements, a predictive interaction modeling module for generating personalized advertisements, and a cross-platform behavior analysis module for analyzing user behavior across offline and online platforms.

A communications system for a restaurant includes multiple field communications units, a portable communications device, and processing circuitry. The field communications units include a speaker and a microphone. The portable communications device is transportable by an order taker of the restaurant. The processing circuitry obtains a user input to transition the portable communications device between multiple different channels. The multiple different channels correspond to one of the multiple field communications units. The processing circuitry transitions the portable communications device to one of the multiple different channels according to the user input. The processing circuitry facilitates end-to-end bi-directional audio communication between the order taker and a customer by operating the portable communications device and the one of the multiple field communications units to exchange audio data over the one of the multiple different channels.

The invention relates to the technical field of earphone noise reduction processing, and discloses a motorcycle riding earphone noise reduction method combining different noise reduction algorithms, comprising the following steps: acquiring an original audio signal, and distributing the original audio signal to each processing module; a preprocessing signal is generated, voice activity information is determined, and environmental noise energy information is calculated; according to the environmental noise energy or the auxiliary information, adaptively adjusting a high-low noise energy threshold value; the original signals are input into the traditional and AI noise reduction module in parallel to generate two paths of noise reduction signals; based on the environment noise, the voice information and the threshold value, performing weighted fusion to generate an output signal; and carrying out equalization and compression processing on the output signal, and driving the loudspeaker to output. According to the invention, comprehensive judgment on environmental noise energy information, voice activity information and riding speed is introduced, an output signal of a traditional noise reduction module is set to be zero in a high-speed voice-free environment, and AI noise reduction is combined, so that tone quality distortion of a traditional noise reduction algorithm in a noise environment is effectively avoided.

The present invention provides a speaker includes a frame, a vibration system fixed with the frame, a magnetic circuit system driving the vibration system to vibrate and sound, a flexible circuit board and an electrical connector. The vibration system includes a diaphragm and a voice coil driving the diaphragm to vibrate. The flexible circuit board is fixed on a side of the diaphragm distal to the voice coil. The flexible circuit board covers and is fixed with the diaphragm. The electrical connector passes through the diaphragm to electrically connect the flexible circuit board and the voice coil. In present invention, the flexible circuit board does not occupy a space of the magnetic circuit system, and the acoustic performance of the speaker is good.

Various implementations include audio systems and approaches for creating personal audio zones. Certain implementations include a system having: a first set of near-field (NF) speakers; a set of sensors for detecting a position of a first user; and a controller coupled with the first set of NF speakers and the set of sensors and configured to adjust an audio output at the set of NF speakers, wherein the controller is configured to maintain spatialization of the audio output to the first user based on detecting at least one of a position change or an orientation change of the first user, and wherein the controller is configured to control the spatialized audio output to the first user with the first set of NF speakers with consideration of isolation to a second listening location.

The embodiment disclose a wearable augmented reality eyewear system including a frame supporting left and right transparent lenses. A plurality of outward-facing cameras mounted on the frame have overlapping fields of view for capturing real-time images of the environment. A plurality of inward-facing cameras integrated in the frame or lenses detect pupil size, pupil dilation, and gaze direction using infrared imaging. A digital processor analyzes gaze direction and fixation based on virtual grid coordinates, initiates an object or person recognition routine using images from the outward-facing cameras, and compares detected features with records in a database. At least one lens includes an AR projection system embedded therein and configured to superimpose data about the recognized object or person in the user's field of view. An audio speaker positioned near the user's ear outputs related audio. A wireless communication module enables real-time data exchange with a cloud-based platform.

The invention discloses an audio data processing method and device, a storage medium and electronic equipment. The method comprises the steps that to-be-processed audio and at least one track point are acquired, initial audio data are generated based on the to-be-processed audio and the track point, loudspeaker gain corresponding to a target virtual loudspeaker is determined based on the initial audio data, and position information of the track point is used for determining a virtual source position in a virtual playing scene. Virtual loudspeakers are deployed in the virtual playing scene, the target virtual loudspeaker is at least one of the virtual loudspeakers, the target virtual loudspeaker represents the virtual loudspeaker determined based on the virtual source position, and the loudspeaker gain is used for simulating playing of the to-be-processed audio at the virtual source position; and performing sound mixing processing on the audio to be processed and the media information indicated by the source media information identifier based on the loudspeaker gain to generate target audio data. The technical problem that the playing effect of the audio data is poor is solved.

The invention provides a sound field optimization method and device of a multichannel loudspeaker array, equipment and a medium, and relates to the technical field of acoustics, to-be-played audio signals are respectively input to each phase compensation filter for phase compensation filtering, and a plurality of phase-compensated audio signals are obtained; the phase compensation filter is determined according to the phase compensation value of the loudspeaker; respectively inputting each phase-compensated audio signal into a corresponding loudspeaker so as to control the multi-channel loudspeaker array to produce sound; acquiring sound field response data in the target area; and when the difference between the sound field response data and the target sound field data does not meet the set sound effect requirement, dynamically adjusting the phase compensation value of the loudspeaker, and repeating the step of phase compensation filtering until the difference between the obtained sound field response data and the target sound field data meets the set sound effect requirement. According to the invention, through a dynamic adjustment mechanism and a closed-loop optimization process, precise regulation and control of the sound field of the loudspeaker array are realized, and the sound effect quality in the target area is significantly improved.

A loudspeaker system and method can measure a specific user's hearing and implement compensatory or corrective processing to address the user's hearing deficiencies. The user's hearing acuity may be determined by (i) headphone techniques in which frequency-based hearing thresholds are determined; (ii) a loudspeaker system set up as intended for use in an acoustic space to emit tonal stimuli; or (iii) inducing, acquiring, and interpreting otoacoustic emissions. Once the hearing profile is determined, appropriate signal processing parameters can be used to generate a corrected audio output signal adjustment. The administered hearing test results can include a plurality of hearing loudness threshold levels at respective test frequencies. The hearing correction system's corrected audio output can be generated from corrected loudness values at selected correction frequency sub-bands.

The invention discloses an external voice changing method and system, and the method comprises the steps: building a mapping relation between a voice changing scene and voice changing parameters, and enabling the voice changing scene to be used for representing an external environment where a vehicle is located; in response to the user operation, determining a target voice changing scene; obtaining a target voice changing parameter corresponding to the target voice changing scene based on the mapping relationship between the voice changing scene and the voice changing parameter; and performing voice changing processing on voice played by a loudspeaker outside the vehicle by adopting the target voice changing parameter. According to the method, the voice interaction mode of the vehicle and the external environment has scene perception and self-adaptive capabilities, and the voice outside the vehicle can be matched with the specific environment and is easier to accept.

A work machine includes a lower traveling body; an upper slewing body slewably mounted on the lower traveling body; an external microphone configured to collect sound around a work machine; an internal speaker configured to output sound to an operator of the work machine; an information transmission device configured to enable a worker around the work machine to recognize whether or not a voice of the worker has reached the operator of the work machine; and a processor, and a memory storing instructions that cause the processor to execute a process. The process includes collecting, by the external microphone, sound around the work machine; outputting, by the internal speaker, the sound to the operator of the work machine; and enabling, by the information transmission device, the worker around the work machine to recognize whether or not a voice of the worker has reached the operator of the work machine.

The invention relates to the technical field of noise reduction seats, in particular to a subway cab active noise reduction system and method based on a multi-channel feedback algorithm. The device comprises a headrest shell, the headrest shell is installed at the top of a seat of a metro cab through a headrest support, a loudspeaker, an error sensor and a master controller are arranged in the headrest shell, and the headrest shell is sequentially provided with a front shell, a rear shell and a long-strip-shaped check block; the headrest supports are correspondingly installed in the long-strip-shaped check blocks on the two sides of the headrest shell respectively and move up and down along the long-strip-shaped check blocks, and the height of the headrest shell is adjusted. The loudspeakers are tightly attached to the ears of a driver at certain intervals; the error sensor forms a rhombic array covering a vertical range of 10cm; and the main controller is fixed on the front shell through the notch of the rear shell. According to the invention, the parameters of the main acquisition filter and the auxiliary acquisition filter of the main controller are adjusted according to the error signal to continuously adapt to the complex and changeable noise environment in the subway cab, so that stable and efficient active noise reduction is realized.

The present disclosure discloses a loudspeaker apparatus. The loudspeaker apparatus may include a support connection member configured to contact with a user's head. The loudspeaker apparatus may include at least one loudspeaker assembly. The loudspeaker assembly may include an earphone core and a core housing configured to accommodate the earphone core. The core housing may be fixedly connected to the support connection member. At least one button module may be arranged on the core housing. The interior of the core housing may further include at least two microphones, and the at least two microphones may be arranged at a position different from a user's mouth. The loudspeaker apparatus may further include a control circuit or a battery accommodated in the support connection member. The control circuit or the battery may drive the earphone core to vibrate to generate sound.

------ Acoustic Enclosure and Method for Leveling an Acoustic Enclosure The present invention relates to an acoustic enclosure (20) comprising a loudspeaker (21) mounted flush on a front face of the enclosure (20), characterized in that at least one Helmholtz resonator (26) is formed on the front face of the enclosure (20) between each lateral edge of the front face and the loudspeaker (21), the at least one Helmholtz resonator (26) being dimensioned to have a band gap to block the propagation, in the plane of the front face of the enclosure (20), of acoustic waves in the useful frequency band of the loudspeaker (21). Figure to be published with the abstract: Figure 6a

Methods for generating object-based audio programs that are renderable in a personalizable manner and include a speaker channel bed that is renderable without selection of other program content (e.g., to provide a default full-range audio experience). Other embodiments include steps of delivering, decoding, and / or rendering such programs. Rendering of the content of the bed or selected mix of other content of the program can provide an immersive experience. The program can include a plurality of object channels (e.g., object channels indicative of user-selectable and user-configurable objects), a speaker channel bed, and other speaker channels. Another aspect is an audio processing unit (e.g., an encoder or decoder) configured to perform any of the embodiments of the method or that includes a buffer memory storing at least one frame (or other segment) of an object-based audio program (or bitstream thereof) generated according to any of the embodiments of the method.

A loudspeaker assembly and a headphone are provided. The headphone includes a housing assembly, a bone-conduction core module, and an air-conduction core module, and an accommodation space. The bone-conduction core module is provided in the accommodation space and vibrates in a first vibration direction, the air-conduction core module is provided in the accommodation space, and the bone-conduction core module and the air-conduction core module are arranged along the first vibration direction and face each other. In the above manner, the present disclosure reduces the influence of the air-conduction core module on the vibration effect of the bone-conduction core module, increases the sound volume, and improves the bone-conduction effect of the loudspeaker assembly, thereby improving the sound quality effect of the loudspeaker assembly.

The invention provides a vibration sounding device which comprises a shell which comprises a top wall, a bottom wall and a side wall, and the top wall is provided with a sounding hole penetrating through the top wall; the sound production single body comprises a basin frame, and a magnetic circuit system and a vibrating diaphragm which are respectively fixed on the basin frame, and the vibrating diaphragm comprises a first sound production part and a second sound production part which are arranged side by side; the sound production monomer further comprises a first voice coil and a second voice coil. The elastic supporting piece is used for elastically suspending the sound production single body in the shell; the elastic sealing ring is arranged around the sound outlet hole; and the stator is a driving coil which is arranged at the first magnetic gap and / or the second magnetic gap and is used for driving the sound production monomer to vibrate. According to the vibration sound production device, a receiver, a loudspeaker and a motor can be integrated, the size is small, private conversation can be achieved in function, meanwhile, the sound production front cavity and the sound production rear cavity of the sound production single body can be blocked, and the effects of preventing water and dust and providing damping are achieved.

The invention discloses a debugging method and system of an audio system, and relates to the technical field of audio debugging, the method comprises the steps of S1, environmental acoustic analysis, S2, basic parameter presetting, S3, real-time monitoring and dynamic adjustment, and S4, adaptive optimizing.The layout of an equalizer or a loudspeaker is adjusted through environmental acoustic analysis; presetting basic parameters of the sound amplifying device and the microphone; real-time multi-band sampling, delay calibration and gain regulation are realized; adaptive optimization is carried out, algorithms such as noise reduction are started, stored parameters are scene presetting, and one-key calling and self-checking updating are supported; the environment is analyzed, the audio system is accurately debugged, the system is dynamically optimized, the tone quality and the debugging efficiency are improved, and diversified scene requirements are met.

The present disclosure provides an earphone, comprising a housing component, a loudspeaker, and a protective net. The housing component is provided with an accommodation cavity and a pressure relief hole. The pressure relief hole is configured to connect the accommodation cavity with an external environment. An opening of the pressure relief hole is disposed at a bottom of the housing component along a height direction. The loudspeaker is disposed in the accommodation cavity. The protective net is connected to the housing component to cover the opening of the pressure relief hole. When projected along the height direction, a length of a projection of the protective net is more than 0.4 times a length of a projection of the housing component.

Methods and apparatus are described herein related to improving the sound quality of a bone conduction speaker. The sound quality of the bone conduction speaker is adjusted in the sound generation, sound transferring, and sound receiving of the bone conduction speaker by designing vibration generation manners and vibration transfer structures.

The invention discloses a double-earphone sound effect adjusting method and system, a storage medium and a program product, and relates to the field of loudspeakers, and the method comprises the steps: collecting an environment audio signal through a binaural microphone, and executing spectrum analysis to obtain a frequency distribution characteristic of the environment audio signal; based on a preset human voice frequency range, extracting an audio signal conforming to the human voice feature from the frequency distribution feature as a target audio; calculating the time difference and the intensity difference of the target audio between the left ear and the right ear to obtain the human sound source direction and the relative distance in the environment; acquiring head attitude sensor data, and calculating a matching included angle between the head orientation of the user and the human sound source direction based on the head attitude sensor data; when the matching included angle is smaller than a preset threshold value, increasing the audio signal gain of the head facing the direction; and when the conversation voice of the user is detected, the audio signal gain is kept unchanged. By implementing the application, the interaction efficiency when the double earphones are used can be improved.

The invention provides an electronic device. The electronic equipment comprises a first screen, a middle frame, a rear cover assembly and a loudspeaker. The first screen and the rear cover assembly are mounted on two opposite sides of the middle frame; the loudspeaker is located between the first screen and the rear cover assembly, the loudspeaker is installed in the middle frame, and a front cavity is defined by the loudspeaker and the rear cover assembly; the electronic equipment is provided with a sound outlet which is communicated with the front cavity; the sound outlet is arranged on the rear cover assembly, or the sound outlet is formed by the rear cover assembly and the middle frame in a surrounding mode. According to the electronic equipment, the sound outlet is formed in the rear cover assembly, or the sound outlet is defined by the rear cover assembly and the middle frame, so that sound output of the rear side of the electronic equipment is achieved, and the sound output mode of the electronic equipment is optimized.