4092results 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.

Multichannel audio playback devices and associated systems and methods are disclosed herein. In some examples, a first playback device is configured to receive a source stream of audio content comprising left, right and center input channels. In a first mode, the first playback device is configured to play back audio via a plurality of transducers based on the left, right, and center input channels. In a second mode, in which the first playback device is bonded to second and third playback devices, the first playback device is configured to (i) play back audio via the plurality of transducers based on at least the center input channel, (ii) cause audio to be played via the second playback device based on at least the right input channel, and (iii) cause audio to be played via the third playback device based on at least the left input channel.

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).

The invention relates to a sound pressure adjusting method and system based on a human body induction sound box. The method comprises the following steps: acquiring user motion data through a human body induction sensor, and performing triangulation positioning analysis on the user motion data through an MCU (Microprogrammed Control Unit) chip to obtain position coordinate data and motion track data; inputting a sound field analysis model in the DSP chip for characteristic analysis to obtain surrounding sound field characteristic parameters; the method comprises the following steps: performing multichannel separation and time domain equalization processing on an original audio signal by a DSP chip to obtain a multichannel preprocessed signal; inputting into a DSP chip for sound pressure balance calculation to obtain an initial sound channel gain control parameter; performing power amplification on the multichannel pre-processed signal, and driving a loudspeaker array to output a surround sound signal; and the DSP chip dynamically optimizes the initial sound channel gain control parameter to obtain a target sound channel gain control parameter. According to the invention, real-time optimization and dynamic updating of sound pressure parameters are realized, and the response speed of the human body induction sound box to environmental changes is improved.

The invention discloses a sound field optimization method and device for a multichannel audio system, equipment and a storage medium. The sound field optimization method comprises the following steps: initializing the system and acquiring acoustic characteristic data; simulating a propagation path of sound waves based on a sound ray tracking algorithm and identifying measurement data, and constructing a 3D sound field model; calculating a TDOA result of a sound channel signal between different microphone positions based on a GCC-PHAT algorithm, obtaining a listening position, combining the TDOA result to calculate an optimal delay value, and performing corresponding compensation; a target mathematical model is formed, a sound field optimization problem of a plurality of listening positions is converted into a multi-target optimization problem, a Pareto optimal solution is generated based on an NSGA-II algorithm, a solution for balancing performance of different positions is selected, the sound quality of other positions is considered while the optimal effect of the main listening position is ensured, the listening experience of a single position is remarkably improved, and the listening experience of the single position is improved. And sound field optimization in a whole space range is realized, and immersive audio experience is brought to the user.

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.

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).

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.

A computer-implemented method for generating crosstalk cancellation (CTC) filters for a reverberant acoustic environment having multiple audio zones, the method comprising determining, based on a CTC simulation model of an acoustic environment, a first set of frequency responses for respective positions proximate to one or more speakers located in the acoustic environment, and a second set of frequency responses for respective listening positions in the acoustic environment, generating, based on the first set of frequency responses and the second set of frequency responses, an acoustic environment transfer function matrix for the acoustic environment, and generating, based on the acoustic environment transfer function matrix, a set of CTC filters for a set of speakers in the acoustic environment.

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.

The embodiment of the invention discloses an audio pairing method and system and a medium, and the method comprises the steps: responding to a condition that a triggering condition is satisfied, entering a pairing mode, and generating pairing information; generating an audio sound wave signal based on the pairing information; and playing the audio sound wave signal through the loudspeaker device so as to send the audio sound wave signal to the to-be-paired device.

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.

Technology for grouping, consolidating, and pairing individual playback devices with network capability (players) to stimulate a multi-channel listening environment is disclosed. An example method includes receiving an audio signal containing a range of audio frequencies; amplifying, according to a gain parameter, the audio signal to be reproduced by at least one speaker. The example method includes automatically increasing the gain parameter to a higher gain parameter responsive to a determination that no more than a subset of the range of audio frequencies is to be reproduced by the at least one speaker. The example method includes amplifying, according to the higher gain parameter, the audio signal containing no more than the subset of the range of audio frequencies to be reproduced by the at least one speaker.

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.

Some disclosed methods involve receiving, by a control system, location control data from a sound source as the sound source emits sound in a plurality of sound source locations within an audio environment. Some such methods involve receiving, by the control system, direction of arrival data from each audio device of a plurality of audio devices in the audio environment. In some examples, each audio device of the plurality of audio devices includes a microphone array and the direction of arrival data corresponding to microphone signals from microphone arrays responsive to sound emitted by the sound source in the plurality of sound source locations. Some such methods involve estimating, by the control system, sound source locations and audio device locations based, at least in part, on the location control data and the direction of arrival data.

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.

Examples described herein relate to management of concurrent audio streams from different sources. Portable playback devices, such as wearable wireless headphones and earbuds, as well as portable battery-powered speakers, may include multiple network interfaces for connection to different types of networks, such as an 802.11-compatible network interface for connection to wireless local area networks (e.g., Wi-Fi® networks) and an 802.15-compatible network interface for connection to a mobile device via a personal area network (Bluetooth®). Via such connections, the playback devices may receive two or more concurrent streams. By managing these streams according to playback policies, the portable playback devices may play the user's intended audio without necessarily requiring user input to explicitly select among the concurrent streams.

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.