617results about "Public address systems" patented technology

A method for synchronizing audio reproduction in collocated end devices is presented. Each of the devices auto-correlates using noise to determine a threshold prior to the antenna receiving an audio signal. When the devices receive a common audio signal, they provide audio outputs. Each device cross-correlates its audio output with the audio outputs of the other devices. The timing of the audio output of each device is then adjusted such that the audio outputs of all of the devices align temporally with the lagging or leading device.

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.

A system, method, and computer-readable medium for performing a smart audio operation. The smart audio operation includes detecting a plurality of audio devices available for use by an information handling system; monitoring information handling system audio related information; analyzing the information handling system audio related information using a trained smart audio artificial intelligence model; and, automatically connecting the information handling system to a particular audio device of the plurality of audio devices based upon the information handling system audio related information.

Methods, apparatus, systems, and articles of manufacture are disclosed to generate spatial audio based on computer vision. An example apparatus includes at least one memory, instructions in the apparatus, and processor circuitry to execute the instructions to determine a position of an audio source based on an image generated via a camera, and apply an audio spatialization filter to an audio signal generated by a microphone based on the position of the audio source.

This disclosure provides methods, devices, and systems for audio signal mixing. The present implementations more specifically relate to mixing audio signals from a microphone array by performing fixed beamforming to generate beams, reducing noise on the beams, and mixing the beams to generate a final audio signal for playback. In some aspects, an audio mixing system includes a fixed beamformer to generate beams from audio signals from a microphone array and noise reduction units (NRUs) to reduce a noise component of each audio beam. The system also includes logic to calculate a signal characteristic of each reduced noise audio beam to determine, based on the signal characteristics, the reduced noise audio beams that include a speech component. The logic also generates a gain for each audio beam based on the selection, with the gains used in beam mixing. In some aspects, the NRU includes a neural network noise reduction unit.

A method for adaptive additive sound includes receiving ambient sound data corresponding to ambient sound in a first zone acquired by a microphone in the first zone, analyzing the ambient sound data from the first zone, generating audio signal data for the second zone based at least in part on the ambient sound data from the first zone, and transmitting the audio signal data for the second zone to a speaker in the second zone. The first zone is separate from the second zone within a space.

Digital signal processing (DSP) may be distributed across a plurality of endpoints and one or more aggregators in an audio system. An aggregator may perform DSP and / or may control DSP features performed by endpoints based on DSP feature capability of the endpoints and the aggregator. Links may be established between the aggregator and the endpoints to communicate audio signals and related information for performing the DSP features.

An audiovisual system uses a spatial position detection module to process audio signals and determine the location and orientation of a speaking participant. Based on this data, the system dynamically controls sensors to optimize audio and video capture. Behavioral and contextual information may also be used to train intelligence models for improved system performance. Further, sensor arrays may be utilized to identify gaze vectors of participants to select a camera sensor of the sensor array. Meeting content collected with the sensor array may be organized into meeting metrics in accordance with an analytics strategy before training an intelligence module with the meeting metrics. Meeting content may also be configured into digital tiles in accordance with a tile strategy. At least one of the digital tiles may be altered in response to a meeting condition detected by the sensor array.

A device is configured to (i) receive media content from a media source, (ii) generate a first series of frames including first portions of the media content and first playback timing information, (iii) generate a second series of frames including second portions of the media content and second playback timing information, (iv) transmit the first series of frames to a first playback device in a synchrony group for playback of the first portions of the media content in accordance with the first playback timing information, and (v) transmit the second series of frames to a second playback device in the synchrony group for playback of the second portions of the media content in accordance with the second playback timing information such that the media content is played back in synchrony by the synchrony group.

A communication protocol provides group communications between at least two wireless headsets. Multiple broadcast links, mutually time staggered, are established between the headsets for sequentially sending broadcast messages between group members. Group members can retransmit and forward one or more received broadcast messages to improve robustness and range. In the receiving headset, audio received in broadcast messages from multiple members is combined into a single audio signal provided to the loudspeaker of the receiving headset.

The invention provides a method, a device and a system for audio signal mixing. The present implementation more particularly relates to mixing audio signals from a microphone array to generate beams by performing fixed beamforming, reducing noise on the beams, and mixing the beams to generate a final audio signal for playback. In some aspects, an audio mixing system includes a fixed beamformer to generate beams from audio signals from a microphone array and a noise reduction unit (NRU) to reduce noise components of each audio beam. The system also includes logic to compute a signal characteristic of each de-noised audio beam to determine a de-noised audio beam including a speech component based on the signal characteristic. The logic also generates a gain for each audio beam based on the selection, wherein the gains are used in beam mixing. In some aspects, the NRU includes a neural network noise reduction unit.

Systems and methods are disclosed in which a playback device transmits a first sound signal including a predetermined waveform. In one example, the playback device receives a second sound signal including at least one reflection of the first sound signal. The second sound signal is processed to determine a location of a person relative to the playback device, and a characteristic of audio reproduction by the playback device is selected, based on the determined location of the person.

Systems and methods for mediating an audio source during a meeting are provided. An active audio source is identified and mapped to instances connected to a communications and collaboration platform to determine a mapped instance. An audio signal device is determined from multiple devices for capturing audio from the active audio source. An audio system is configured to manipulate an active audio signal from the audio signal device via the mapped instance. In some cases, upon identifying a new active audio source, a new mapped instance is determined corresponding to the new active audio source.

The invention provides a sound amplification device based on an M-H metamaterial and a sound transmission method using the sound amplification device, the sound amplification device at least comprises an upper shell, a lower shell and a waveguide located between the upper shell and the lower shell, and the waveguide is made of the Mie metamaterial; the waveguide comprises an outer wall, a plurality of sound wave inlets are formed in the outer wall, and the upper shell, the lower shell and the outer wall enclose a waveguide space; a through hole penetrating through the waveguide and the upper shell is formed in the waveguide space, and the through hole is a sound wave outlet in the sound amplification device; the waveguide comprises a plurality of fan parts which are arranged around the through hole in a circumferential radiation manner; a waveguide channel penetrating through the sound wave outlet and the sound wave inlet is formed between the adjacent fan parts; the control current of the sound source is determined by the size of the sound amplifying device and the scattering distribution condition of the sound amplifying device in different resonance modes, the sound at the sound wave outlet is larger than the sound at the sound source in energy level, and an included angle exists in the propagation direction. By determining the control current at the sound source which is most matched with the size of the sound amplifying device, the sound propagation control with low cost, high accuracy and wide application range is realized.

In a wireless stereo system including first and second battery-powered speakers, if low power is sensed in the second speaker, the user may be given the option of shifting the first speaker to mono mode, or maintain the first speaker in stereo mode, playing its assigned channel, or alter a left / right stereo balance between the speakers. Speaker setting may also change depending on whether a speaker is on or off a charging dock.

Generative media content (e.g., generative audio) can be played back across multiple playback devices concurrently. A generative content group coordinator device can receive input parameters, which can include sensor data, media content, or other such input. The coordinator device can generate first and second generative media content streams, each of which can be transmitted to first and second playback devices, respectively. The first and second playback devices can play back the first and second streams of generative media content concurrently.

Disclosed embodiments relate to Multi-Player Playback Devices. In some embodiments, the Multi-Player Playback Device is configured to selectively operate in one of a plurality of operating modes. In each operating mode, the Multi-Player Playback Device is configured to implement a multi-stage audio processing procedure comprising: (i) routing incoming audio streams to mixer inputs of a mixer based on a current operating mode in which the Multi-Player Playback Device is operating; (ii) for one or more audio streams routed to an individual mixer input, (a) generating a mixed stream based on the one or more audio streams routed to the mixer input, and (b) routing the mixed stream to at least one audio output of the Multi-Player Playback Device based at least in part on a channel map associated with the current operating mode in which the Multi-Player Playback Device is operating.

The utility model relates to the technical field of loudspeakers, and particularly discloses an invisible sound reinforcement plate which comprises a middle core layer which is a composite base material formed by interweaving carbon fibers and aramid fibers. The supporting layer is arranged at the upper end of the middle core layer, and the supporting layer is composed of a plurality of frameworks with the sections of regular hexagon structures and is of a honeycomb structure as a whole; the sound reinforcement plate has the advantages of high strength, good tensile effect, support of ultrahigh-frequency sound wave conduction, light weight and reduction of the overall weight of the sound reinforcement plate; the supporting layer is composed of a plurality of frameworks with the sections of regular hexagon structures, the whole supporting layer is of a honeycomb structure, rigidity is good, bending resistance is high, deformation is not prone to being caused in the assembling and using process, and therefore the using performance of the sound reinforcement plate is improved. The supporting ribs form a continuous force transmission path and disperse local stress, the trapezoid cross section reduces the stress concentration coefficient, and the buffering wrapping edge is made of silica gel materials, prevents external dust from invading the sound reinforcement plate interlayer and meanwhile plays a role in buffering and shock resistance.

Embodiments are provided for populating a preference database affiliated with a media playback system with preferences received for media items. A preference may be associated with a media item from a service provider account, and may be stored in the preference database in association with the service provider account. The service provider account may be associated with multiple user profiles used to access a media playback device, and the preference may be associated with one of the user profiles associated with the service provider account, and stored in association with the user profile. Further, the preference may be received from a user account different from a user account used to access the media item. In this case, the preference may be stored in association with the user account providing the preference. The stored preferences may be used to manage media content playback by the media playback system.

Systems and techniques for localization of disturbances are illustrated. One embodiment includes a method for detecting a disturbance in an area of operation, the area of operation including a plurality of transmitter-receiver pairs. The method obtains, based on a first plurality of signal transmissions between the plurality of transmitter-receiver pairs, at least one baseline channel impulse response (CIR) measurement. The method obtains, based on a second plurality of signal transmissions between the plurality of transmitter-receiver pairs, at least one additional CIR measurement; The method obtains, based on the at least one additional CIR measurement and the at least one baseline CIR measurement, at least one residual CIR measurement. The method derives, based on a sufficient similarity between the at least one residual CIR measurement and at least one particular key of a plurality of keys, localization data for at least one disturbance detected within the area of operation.

Example techniques may involve multiple calibrations for one or more playback devices. An example implementation may involve detecting, via a microphone, calibration sounds as emitted by one or more playback devices during a calibration sequence, perhaps by recording first samples while the microphone is in motion through a given environment and recording second samples while the microphone is stationary at one or more particular locations. The implementation may also include determining a first calibration for the one or more playback devices based on at least the first samples of the calibrations sounds and determining a second calibration for the one or more playback devices based on at least the second samples of the calibrations sounds. The implementation may further include applying at least one of (a) the first calibration or (b) the second calibration to playback by the one or more playback devices.

The present invention provides a method for processing music audio data, comprising the steps of providing input audio data representing a first piece of music containing a mixture of predetermined musical timbres, decomposing the input audio data to generate at least a first audio track representing a first musical timbre selected from the predetermined musical timbres, and a second audio track representing a second musical timbre selected from the predetermined musical timbres, applying a predetermined first audio effect to the first audio track, applying no audio effect or a predetermined second audio effect, which is different from the first audio effect, to the second audio track, and obtaining recombined audio data by recombining the first audio track with the second audio track.

An apparatus comprises receivers (101, 102) receiving microphone signals and a video signal capturing a scene. An image generator (113) generates an image signal comprising images generated from the video signal and a zoom determiner (115) determines a zoom property, such as a zoom factor, for the images. An audio component generator (107) generates a stereo signal component for a given mono audio signal to comprise the mono audio signal positioned at a position in a stereo image of the stereo signal component. A combiner (109) generates the stereo audio signal to include at least one stereo signal component and an output (111) generates the audiovisual signal to comprise the stereo audio signal and the image signal. In the approach the position is determined in dependence on the zoom property, and typically also dependent on a direction of arrival for the given mono audio signal. The approach may allow closer consistency between visual and auditory spatial cues.

The invention provides a multi-room audio distribution method and system based on voice recognition, and the method comprises the steps: receiving a voice instruction of a user, and converting the voice instruction into text information; semantic analysis and intention recognition are conducted on the text information, and specific operation instructions and operation parameters are extracted; the operation instructions and the operation parameters are distributed to at least one function module related to the multi-room audio system, and the function module comprises an input source selection module, an input source volume control module, a multi-path output volume control module, a multi-path input and output routing module and a sound mixing module; the at least one function module executes corresponding operation according to the received operation instruction and the operation parameter so as to realize distribution of multi-room audios; and generating voice feedback information corresponding to the operation, and returning the voice feedback information to the user.

An over-ear wearable device includes a housing with a first portion configured to be worn behind an ear of a user and a second portion configured to extend over the ear to position an output near an ear canal of the user. A memory is disposed within the housing and stores map data for an airport. A navigation sensor is disposed within the housing and configured to generate location data. A multi-button user input interface is coupled to the first portion. A processor is disposed within the housing and configured to responsive to selection of a first button of the multi-button user input interface, iterate through a list of destinations identified in the map data, responsive to selection of a second button of the multi-button user input interface, designate a target destination, and generate verbal navigation instructions based on the location data, the map data, and the target destination.

Example techniques relate to playback based on acoustic signals in a system including a first network device and a second network device. A first network device may detect a presence of a user using a camera and / or infrared sensors. The first network device sends, in response to detecting the presence of the user, a particular signal via the first network interface. The second network device receives data corresponding to the particular signal and plays back an audio output corresponding to the particular signal.

Embodiments are provided for populating a preference database affiliated with a media playback system with preferences received for media items. A preference may be associated with a media item from a service provider account, and may be stored in the preference database in association with the service provider account. The service provider account may be associated with multiple user profiles used to access a media playback device, and the preference may be associated with one of the user profiles associated with the service provider account, and stored in association with the user profile. Further, the preference may be received from a user account different from a user account used to access the media item. In this case, the preference may be stored in association with the user account providing the preference. The stored preferences may be used to manage media content playback by the media playback system.

A computing apparatus with the proposed Method HRTF- for soundstage extension, upon which some inherent issues of the conventional Method HRTF such as the need for pre-measuring or simulating, highly individual dependence, small sweet spot, etc., have been effectively resolved. With the approaches that decoupling the individual-dependent segment from the entire HRTF chain, moving the intersection-point forward outside the ear, and simplifying the frequency response with relative-quantity approximations, the goal- "listening with one's own ear" is achieved with robustness improved and sweet spot enlarged. The present invention can be deployed in diverse scenarios: the stereo speakers; a foundational algorithmic module for advanced applications such as home theater systems, etc.; and an audio file in a particular encoding format can be also generated upon the invention that the soundstage can be extended readily through replaying with conventional stereo speakers.

The invention relates, inter alia, to a method for sound reinforcement of at least one audience area (3) by means of at least one sound transducer arrangement (1) with a plurality of sound transducers (9), wherein the individual sound transducers (9) of the at least one sound transducer arrangement (1) each emit elementary waves (8) which superimpose to form a common wavefront (4), characterized in that a) the at least one sound transducer arrangement (1) and the at least one audience area (3) are geometrically linked to each other by a coordinate system (2) and b) a spatial assignment exists between the physical positions of the individual sound transducers (9) in the at least one sound transducer arrangement (1) and position vectors si for defining coordinates in the area of ​​the at least one sound transducer arrangement (1),and furthermore c) an assignment of points of the coordinate system (2) to points in the at least one audience area (5) according to a position vector ri, wherein d) direction vectors, in particular normalized direction vectors (61) d^i=ri−siri−si, result in the coordinate system (2) and wherein e) depending on the spatial assignment of the position vectors si and the sound transducers (9), delay times τj for the sound transducers (1) are determined, with which elementary waves (8) are radiated by the sound transducers (9), wherein e) the delay times τj of the sound transducers (9) are each chosen such that the local direction (50) of the common wavefront (4) corresponds to the direction of the direction vector, in particular the normalized direction vector (61) d̂i.