2757results about "Loudspeaker signals distribution" patented technology

The present disclosure provides an earphone comprising a housing assembly, a speaker, and an ear hook assembly. The housing assembly includes a first accommodating cavity and a connecting portion that is provided with a socket extending along a first direction. The first accommodating cavity and a first channel and a second channel included in the socket are sequentially connected. A first abutment surface is disposed at transition between the first and second channels. The first abutment surface is inclined to the first direction. The ear hook assembly includes a first connecting end that includes a plug portion. The plug portion is inserted into the socket and includes a second abutment surface and an annular surface. The second abutment surface is inclined to the first direction and faces the first abutment surface. The annular surface extends from an inner edge of the second abutment surface toward the first abutment surface.

An acoustic output device is provided. The acoustic output device includes a housing; a bone conduction vibrator configured to generate bone conduction sound waves and transmit the bone conduction sound waves to a cochlea through the housing to produce sound; an air conduction vibrator configured to generate air conduction sound waves, the air conduction sound waves being transmitted to an ear through sound guiding holes in the housing; and a processing module configured to provide the bone conduction vibrator and the air conduction vibrator with a first audio signal and a second audio signal, respectively, the first audio signal and the second audio signal having a frequency division point, the first audio signal and the second audio signal including components with frequencies above and below the frequency division point, respectively. The frequency division point is not less than 600 Hz.

A computer-implemented system and method for pupil-directed retinal projection in near-eye displays are disclosed. The computer-implemented system provides smart glasses with directed physical pixels that project light beams / signals directly onto a user's retina based on pupil position and size. The glasses comprise a frame, lenses with directed pixel layers, and sensors for tracking pupil movement. Each directed pixel may generate multiple virtual pixels by rapidly changing its emission angle. The glasses function as prescription lenses, virtual reality displays, and augmented reality devices without traditional optical systems. Additional features include depth sensors, cameras, and connectivity to peripheral devices. The glasses enable a seamless blend of virtual and real-world experiences, creating an immersive “Mixverse” environment. Various input methods, including gesture recognition and brain-computer interfaces, allow for intuitive control and interaction.

Disclosed herein is an audio system that can be customized by the user (including the developer) and that may allow the user to control more than just the sound levels and may allow the user to select more than one of a few pre-determined settings. Disclosed herein is a development platform that allows manufacturers (including the developers) to test various different parameters, settings, algorithms, functions, processes, accessories, and the like. The development platform allows the manufacturer to simulate the performances from possible options for the audio system, and then load the selected option from the development platform to the audio system. Additionally, the development platform that enables the end user to have the same level of control and ability to access, upload, and control plugins on the end user's device(s).

A mixing device is provided. The mixing device includes: an input unit, inputting sounds; multiple output units, outputting the sounds; a sound setting operator, setting a sound setting value that is a setting value relating to the sounds; a selection operator, selecting the output unit of a target for setting the sound setting value; a setting value acquisition means, acquiring the sound setting value set by the sound setting operator with respect to the output unit selected by the selection operator; and a mixing means, for each of the output units, mixing the sounds input at the input unit based on the sound setting value corresponding to the output unit and being the sound setting value acquired by the setting value acquisition means.

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).

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 invention provides a cockpit multi-tone area management method, a related device and a communication system. The cabin system may angry the plurality of screens into a plurality of ride zones according to a profile. Each riding area can be associated with a sound area. The cabin system can dynamically create and delete the sound area according to the access and removal of the sounding device. The cabin system can perform unified management on audio focus, volume, audio routing and media session in a sound area. The application corresponding to one driving area can apply for the audio focus in the sound area associated with the driving area, and the sound production equipment in the sound area is used for playing the audio. Audio playing in different sound areas can be independent and does not affect each other. Therefore, different users in the vehicle can use different sound areas to listen to the audio conveniently, and the audio listening experience of the users in the vehicle is improved.

Turntable control systems are disclosed. A playback device can include an analog audio source component such as a turntable. The playback device is configured to cause playback of first audio based on analog audio signals generated by the analog audio source component. Based on an indication that analog audio signals are no longer being generated, the playback device requests, via a network interface, audio content from one or more remote computing devices. The request can include identifying related content based on the first audio. The playback device receives, via the network interface, second audio content from, and causes playback of second audio based on the second audio content.

A conference host output system disclosed by the present application comprises a main conference room device and a plurality of secondary conference room devices, the main conference room device comprises a conference host, a PoE switch, a plurality of microphone units, a plurality of power amplifiers and a plurality of sound boxes, the secondary conference room devices do not comprise the conference host, and the conference host communicates with each secondary conference room device through a network. The conference host comprises a host control module and an audio control module, each partition output by the conference host is connected with a power amplifier and a sound box, when the partition further comprises at least one microphone unit, the audio control module is connected with the microphone unit through an interface of the PoE switch, and the audio control module is connected with the sound box through the power amplifier. Therefore, the conference host distributes and outputs the plurality of audio input signals to different partitions, an audio processor does not need to be additionally added, an n * n audio matrix function is realized, and a plurality of conference rooms share one conference host, so that the cost of audio equipment is reduced, and the cost performance of a conference audio system is improved.

The invention relates to an audio control system and method based on a multi-area framework and a vehicle, the system comprises a central controller and a plurality of area controllers connected with the central controller, the central controller comprises a first Ethernet module, and each area controller comprises a second Ethernet module, a first microcontroller and a power amplifier driving module. The central controller is configured to carry out data protocol packaging on the first audio data and then transmit a first Ethernet protocol data packet to each area controller. The first microcontroller is configured to receive a first Ethernet protocol data packet transmitted by the central controller, analyze the first Ethernet protocol data packet, and transmit first audio data to the power amplifier driving module. The power amplifier driving module is configured to amplify the first audio data and then drive the audio output device to output the amplified first audio data. The problems that an existing audio control system is complex in communication link, too long in wire harness and the like can be effectively solved.

An audio system receives sounds from a local area with a set of sensors. The audio system analyzes the sounds to identify an amount of background noise in the environment. The audio system may also identify target sound in the environment, such as voice from a nearby speaker, to reinforce. The amount of background noise is then used to adaptively set a level of reinforcement for the target sound. In low-noise environments, the target sound may be minimally, or not reinforced, while in high-noise environments a reinforcement sound signal may be generated with the target sound so that a transducer array may generate reinforcing sounds for the target sound and increase its contrast with the noise in the environment. By applying the reinforcement dynamically, the system automatically adjusts for different sounds and environments.

A method performed by a first device that has an internal clock. The method transmits a signal to a second device to share a clock of the first device. The method presents an extended reality (XR) environment and plays back audio content on the first electronic device, wherein the XR environment is also presented and the audio content is played back on the second electronic device The method generates timebase information that includes a first timebase that that defines a relationship between the shared clock and internal clock and a second timebase that defines a relationship between the first timebase and a playback state of the audio content. The method transmits the timebase information to the second device that synchronizes the playback state of the audio content in the second device, with the playback state of the audio content in the first device.

An audio session management method for an audio environment having multiple audio devices may involve receiving, from a first device implementing a first application and by a device implementing an audio session manager, a first route initiation request to initiate a first route for a first audio session. The first route initiation request may indicate a first audio source and a first audio environment destination. The first audio environment destination may correspond with at least a first person in the audio environment, but in some instances will not indicate an audio device. The method may involve establishing a first route corresponding to the first route initiation request. Establishing the first route may involve determining a first location of at least the first person in the audio environment, determining at least one audio device for a first stage of the first audio session and initiating or scheduling the first audio session.

The invention relates to the technical field of audio equipment configuration, and discloses a parameter configuration method and device of audio equipment, equipment and a storage medium, which are used for improving the accuracy of parameter configuration of the audio equipment. The parameter configuration method of the audio equipment comprises the following steps: acquiring the type and audio format information of a current input signal source; based on the type of the input signal source and the audio format information, identifying an audio content type and determining an optimal playing mode; adjusting a sound box combination and sound channel configuration of the audio equipment based on the optimal playing mode; and matching a volume level according to the audio content type, and carrying out adaptive volume adjustment on the audio equipment in combination with user preference and environment analysis.

A communication apparatus, a synchronization method, and a system are provided. The communication apparatus includes a wireless communication unit, a clock circuit, and a control module. When an electronic device in which the communication apparatus is located is a primary device, the wireless communication unit is configured to send a signal used for synchronization of a communication clock signal. When an electronic device in which the communication apparatus is located is a secondary device, the wireless communication unit is configured to receive the signal that is used for synchronization of a communication clock signal and that is sent by the primary device. The clock circuit may generate a data processing clock signal based on the communication clock signal. The control module may perform data processing and / or control a data output based on the data processing clock signal.

Turntable control systems are disclosed. A playback device can include an analog audio source component such as a turntable. The playback device is configured to cause playback of first audio based on analog audio signals generated by the analog audio source component. Based on an indication that analog audio signals are no longer being generated, the playback device requests, via a network interface, audio content from one or more remote computing devices. The request can include identifying related content based on the first audio. The playback device receives, via the network interface, second audio content from, and causes playback of second audio based on the second audio content.

The invention discloses a control method and device of an audio processing system and a storage medium, and relates to the technical field of audio data processing, and the method comprises the steps: obtaining environment information and device state data collected by a sensor, and inputting the environment information and the device state data into a clock compensation module; the method comprises the following steps: generating a corresponding compensation parameter according to a preset sampling rate, performing a compensation action on an audio clock based on the compensation parameter, determining a source sampling rate corresponding to at least one audio source according to the compensated audio clock, finally detecting the source sampling rate through a sampling rate conversion module, and when the source sampling rate is different from a preset sampling rate, outputting the audio source. And adjusting the source sampling rate based on the preset sampling rate. According to the application, the audio clock is compensated, and the compensated sampling rate is uniformly converted through the sampling rate conversion module, so that clock synchronization among multi-source audio input and sampling rate unification of the multi-source audio input are realized, and the processing efficiency of the audio data is improved.

A wearable audio output device (e.g., headphones) having an open design that allows ambient noise to pass to a listener without physically isolating the listener from a surrounding environment. The device may include an open earcup design that may partially or completely surround the listener's ear, and in some examples a portion of the listener's head may be uncovered by the open earcup. To improve comfort, the device includes a floating audio component configured to generate output audio in a direction of the listener's ear without contacting the listener's ear. To reduce audio leakage into the environment, the floating audio component may include two audio transducers and perform beamforming to target the output audio toward the listener's ear and cancel the output audio in other directions. For example, the beamforming may cause constructive interference in one direction and destructive interference in all other directions.

The invention provides a control method and device of a vehicle-mounted external loudspeaker and a vehicle-mounted external loudspeaker system. The control method comprises the following steps that in response to starting of a vehicle-mounted loudspeaker box mode, the relative position relation between a user and a vehicle is obtained at a preset time interval, and the vehicle is provided with a plurality of vehicle-mounted external loudspeakers making sounds outwards; determining a target loudspeaker and a corresponding sound channel from the plurality of vehicle-mounted external loudspeakers at least based on the relative position relationship; and controlling each target loudspeaker to play the audio data of the corresponding sound channel. The invention provides a mode of taking the vehicle as a large mobile sound box capable of providing a surround sound field, and the provided surround sound field can be adjusted according to the relative position relationship between the user and the vehicle, so that the hearing effect of the user when using the vehicle-mounted sound box is improved, the application scene of the vehicle is expanded, and the user experience is improved.

An audio device comprising, an interface, one or more first output transducers, one or more input transducers, and a processor. The audio device is configured to emit a first ultrasound signal, receive a first reflection indicative of the first ultrasound signal, process the first reflection to determine movement, and in response to movement being determined, adjust one or more audio processing parameters of the audio device.

A battery electric vehicle includes a speaker including an in-vehicle speaker that outputs a sound to the inside of the vehicle and an outside speaker that outputs sound to the outside of the vehicle. The battery electric vehicle further includes one or more processors configured to generate a pseudo engine sound and output the pseudo engine sound from a speaker mounted on the battery electric vehicle. When a vehicle speed of the battery electric vehicle is higher than a first speed, the one or more processors output the pseudo engine sound from the in-vehicle speaker, but not from the outside speaker. When the vehicle speed of the battery electric vehicle is equal to or lower than the first speed, the one or more processors output the pseudo engine sound from both the in-vehicle speaker and the outside speaker.

Example techniques relate to primary-ambient playback of surround audio by audio playback devices. Example playback devices described herein may include multiple speakers, such as a forward-firing audio transducer and side-firing audio transducers. Using example techniques, such playback devices may perform primary-ambient decomposition to separate surround channels into primary′ and ambient channels. The playback devices play back the ambient channel(s) via the side-firing transducers and the primary channel via the forward-firing transducer.

A display device includes a controller that obtains a plurality of sound sources respectively corresponding to a plurality of content, and generates one or more sink sound sources for transmission to at least one of a plurality of sink devices based on one or more sound sources among the plurality of sound sources. The controller may generate one or more sink sound sources for transmission to at least one sink device corresponding to at least one sink device item selected based on a user input for selecting at least one of a plurality of sink device items included in each of a plurality of sink device lists.

A sound system capable of reducing the number of speakers required for performing localized reproduction in stereo is provided. A first speaker emits a sound based on an L (2n−1)-th processed acoustic signal with signal processing to make it audible in a first region in an L direction to an L (2n−1)-th acoustic signal and an R (2n)-th processed acoustic signal with signal processing to make it non-audible in a fourth region in an R direction to an R (2n)-th acoustic signal, and a second speaker emits a sound based on an L (2n)-th processed acoustic signal with signal processing to make it non-audible in a second region in an L direction to an L (2n)-th acoustic signal and an R (2n−1)-th processed acoustic signal with signal processing to make it audible in a third region in an R direction to an R (2n−1)-th acoustic signal.

The invention relates to the technical field of vehicles, in particular to an audio arbitration interaction method and device, a vehicle and a storage medium, and the method comprises the steps: receiving an audio focus request of at least one target audio system and an audio focus of at least one other audio system in a use state; and identifying audio priorities of each target audio system and at least one other audio system in the use state, generating an audio priority identification result, if the highest audio priority exists in the at least one target audio system, allocating a target audio focus to the target audio system corresponding to the highest audio priority, and if the highest audio priority exists in the at least one target audio system, allocating a target audio focus to the target audio system corresponding to the highest audio priority. And adding the target audio focus into a focus stack, and removing the preempted audio focus of the at least one other audio system at the same time, so as to control the target audio system to perform audio playing based on the target audio focus. Therefore, the problems of audio focus conflict, asynchronous audio focus distribution, incapability of determining recovery of the audio focus after the audio focus is preempted and the like in the prior art are solved.

An electronic device including at least one speaker, a communication interface, and at least one processor that is connected with the at least one speaker and the communication interface and controls the electronic device. The processor, when the electronic device is in a first mode, may output one or more first channel signals included in a sound via at least one speaker and control the communication interface to transmit one or more second channel signals in the sound to at least one external speaker, and when the electronic device is in a second mode, mix at least one of the one or more first channel signals and the one or more second channel signals to thereby produce a mixed signal and control the communication interface to transmit the mixed signal to the at least one external speaker.

The embodiment of the invention provides an acoustic device and a signal processing method, a bone conduction sounding assembly of the acoustic device generates a first time delay when converting a first driving signal into a bone conduction sound wave, and an air conduction sounding assembly generates a second time delay when converting a second driving signal into an air conduction sound wave. The absolute value of the difference between the second time delay and the first time delay is larger than 100 microseconds, the signal processing circuit obtains an audio signal during operation, generates a first driving signal based on a first component of the audio signal and sends the first driving signal to the bone conduction sounding assembly, and generates a second driving signal based on a second component of the audio signal and sends the second driving signal to the air conduction sounding assembly, one of the first driving signal and the second driving signal is transmitted in a delayed manner relative to the other, so that the time difference between the generation time of the bone-guided sound wave and the generation time of the air-guided sound wave at at least partial frequency is less than or equal to 100 microseconds. According to the scheme, the problem of partial sound can be avoided.

The invention discloses an automatic audio delay calibration method, device and equipment based on anomaly detection and a storage medium, which are applied to a multi-equipment audio system and are used for collecting audio signals for preprocessing, collecting delay time measurement values, environment data and the audio signals for abnormal data detection. Calculating a delay compensation based on the audio signal and the delay time measurement, calculating an environmental factor compensation based on the environmental data; based on the delay compensation and the environmental factor compensation, audio delay automatic calibration is carried out through a real-time feedback mechanism; and evaluating the system performance based on the audio signal acquired in real time and the delay time measurement value, and optimizing the real-time feedback mechanism in real time. Interference data are accurately identified and filtered through intelligent data analysis, and a real-time analysis and automatic processing mechanism is adopted, so that the limitation of traditional manual calibration is broken through. The system can dynamically sense environment changes and adaptively adjust calibration parameters, and millisecond-level delay calibration precision is achieved.