3489results about "Stereophonic systems" patented technology

A device capable of using image data for purposes of determining a location of a user and audio beam selection to isolate audio in the direction of the user. The beamforming / beam-steering may occur after determining the user's location in order to conserve computing resources that would otherwise have been spent determining beams for non-desired directions. The beamformed audio may be used for speech processing, a communication session involving the device, or other purposes.

Some examples are directed to systems and methods for presenting spatial audio at a plurality of locations based upon movement of a user viewpoint. Some examples are directed to systems and methods for presenting spatial audio with a level of detail based upon a distance between a location corresponding to spatial audio and a user viewpoint. Some examples are directed to systems and methods for changing a level of audio of an audio component associated with a virtual environment in response to a request to change a level of immersion of the virtual environment.

A method of audio processing combining WiFi and BLE technologies includes: obtaining target audio data including multi-channel information; establishing connections among audio devices corresponding to multiple rooms through WiFi and BLE Auracast™ protocols; dynamically configuring distribution paths according to intelligent distance detection results, and obtaining a configured audio distribution network; adjusting, using a synchronization algorithm in the BLE Auracast™ protocol, a timing sequence of the target audio data, and obtaining synchronized audio data; and transmitting, according to the configured audio distribution network, the synchronized audio data via WiFi or the BLE Auracast™ protocol to each audio device for playback.

The invention is applicable to the field of intelligent audio adjustment, and provides an immersive audio-video follow-up adjustment method and system, and the method comprises the steps: constructing a multi-dimensional perception system, and collecting multi-source information in real time; carrying out fusion processing on the collected multi-source information based on a deep neural network model, mining a dynamic mapping relation between a user state and the video content through space-time correlation analysis, and identifying a user interaction intention and an emotional tone and a space scene attribute of the video content; according to a fusion processing result, calling a dynamic parameter adjustment engine, and generating an audio parameter adjustment scheme in real time; a user experience feedback closed loop is constructed, a visual attention area of a user is collected through eye movement tracking equipment, and personalized adjustment preference parameters are generated; according to the method and the device, the audio effect is accurately matched with the user state and the audio and video content, the naturalness and the adaptability of immersive experience are remarkably improved, universality and individual differences are considered, and the audio experience which is more suitable for scenes and needs of the user 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.

Aspects of embodiments pertain to a method for determining an Input / Output (I / O) device configuration for a music teaching system. The method may comprise receiving a plurality of I / O device configurations of a music teaching system; receiving, for a given I / O device of the music teaching system, an initial I / O device configuration; and determining an updated I / O device configuration for the given I / O Device, based on the initial I / O device configuration of the given I / O Device and at least one of the plurality of received I / O device configurations. The determining is performed such that the updated I / O device configuration has improved device performance compared to the initial I / O device configuration.

The invention relates to the technical field of audio processing, in particular to a self-adaptive scene sound effect adjusting method and device and a storage medium, and the method comprises the steps: collecting a mixed audio stream and an original playing audio, and extracting an acoustic fingerprint vector through a pre-trained convolutional neural network; performing acoustic feature deconstruction on the acoustic fingerprint vector to obtain a plurality of acoustic feature components, performing scene recognition judgment according to a preset scene judgment rule based on the plurality of acoustic feature components, and determining an audio playing scene; constructing a time-frequency mask according to the audio playing scene, generating a noise reduction gain matrix based on the time-frequency mask, and performing scene sound field enhancement on the audio playing scene to obtain a sound field optimization matrix; and performing noise reduction and sound effect adjustment on the original playing audio according to the noise reduction gain matrix and the sound field optimization matrix, and outputting the optimized playing audio. According to the invention, sound effect adjustment can be dynamically adapted according to different scenes, and the noise reduction accuracy and the multi-scene sound quality adaptability are 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 relates to the technical field of audio signal processing, and discloses a sound box and sound effect intelligent adjusting method and system based on data analysis and a storage medium, and the sound box and sound effect intelligent adjusting method based on data analysis comprises the steps: constructing a user feature modeling engine, and extracting user auditory characteristic data; constructing an environment characteristic analysis engine, and collecting environment acoustic characteristic data; constructing a content feature extraction engine, and analyzing audio content semantic data; constructing a three-dimensional fusion optimizer, inputting the three feature vectors into an auditory scene fusion model, calculating an auditory experience score through tensor fusion operation, and solving an optimal sound effect parameter by applying a multi-objective optimization algorithm; constructing a parameter generation controller, and converting the optimal sound effect parameter into a specific audio processing parameter; according to the invention, the problem of mutual interference caused by traditional separated processing is solved, and the accuracy of sound effect adjustment and the user satisfaction are improved.

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.

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.

The invention relates to a directional speaker system for a passenger transport system. A passenger transport system (100; 200); 300); 400, 400; 500); 600) comprises: a directional speaker system (106; 206, 206; 306; 406a-c, 406a-c; 506a-b, 506a-b; 606), the directional speaker system being configured to transmit an audio communication (120a-b; 220 a-b; 220 a-b; 320a-b; 320a-b; 420a-e) to a restricted target area (110a-b; 310a-b, 310a-b; 410a-b, 410a-b; 610a-c) in the formula (I). The use of a directional speaker system allows audio communications to be sent to a user (or to multiple users) without the need to broadcast messages more widely. Thus, other users of the system are not disturbed or distracted by messages that they are not interested in or are not related to.

Methods and audio processing units for generating an object based audio program including conditional rendering metadata corresponding to at least one object channel of the program, where the conditional rendering metadata is indicative of at least one rendering constraint, based on playback speaker array configuration, which applies to each corresponding object channel, and methods for rendering audio content determined by such a program, including by rendering content of at least one audio channel of the program in a manner compliant with each applicable rendering constraint in response to at least some of the conditional rendering metadata. Rendering of a selected mix of content of the program may provide an immersive experience.

A passenger conveyance system (100; 200; 300; 400; 500; 600) comprising: a directional speaker system (106; 206; 306; 406a-c; 506a-b; 606) configured to transmit an audio communication (120a-b; 220a-b; 320a-b; 420a-e) into a restricted target area (110a-b; 310a-b; 410a-b; 610a-c). Using a directional speaker system allows audio communications to be sent to a user (or to multiple users) without broadcasting the message more widely. Thus other users of the system are not disturbed or distracted by messages that are not of interest to them or not relevant to them.

Aspects of the subject technology provide individual volume controls for concurrently operating audio sources at an electronic device. The audio sources may be spatialized audio sources that are associated with display objects that are displayed to appear at various three-dimensional locations around a user of the electronic device, such as in an extended reality environment. In this way, an electronic device may provide the user with the ability to individually control the volumes of the various audio streams originating from the various three-dimensional locations around the user. The individual volume controls may be applied according to individual volume control curves, and / or based on a user intent determined by the electronic device.

In aspects of adjusting audio output in a vehicle, a vehicle audio system implements an audio playback manager that detects a location of a first person in the vehicle, the first person located in proximity of a first speaker device configured for audio output. The audio playback manager also detects an additional location of a second person in the vehicle, the second person located in proximity of a second speaker device configured for the audio output. The audio playback manager determines a preference for the first person related to the audio output and an additional preference for the second person related to the audio output. Based on the preference of the first person or the additional preference of the second person, the audio playback manager adjusts a volume of at least one of the first speaker device or the second speaker device.

Disclosed herein are systems and methods for power transmission between playback devices, systems and methods for energy harvesting and distribution for audio playback devices, and systems and methods for wirelessly powering wearable audio playback devices.

Apparatus for retrieving an acoustic room information from a database, comprising: a test fingerprint processor (10) for deriving a test fingerprint from a test acoustic room information; and a database processor (20) for retrieving a reference acoustic room information having a reference fingerprint matching with the test fingerprint, wherein the test fingerprint processor (10) comprises; a filter bank configured for decomposing the test acoustic room information into a plurality of acoustic room information sub-bands; and a parameter calculator (12) configured for calculating a sub-band parameter for each acoustic room information sub-band of the plurality of acoustic room information sub-bands, the sub-band parameter indicating an energy change over time in the acoustic room information sub-band, wherein the test fingerprint is based on the sub-band parameters for the acoustic room information sub-bands.

The present application concerns early reflection processing concepts for auralization. Embodiments relate to apparatuses and methods for sound rendering considering early reflections and to apparatuses and methods for determining an early reflection pattern.

A computer-implemented sound calibration method, a system and a storage medium are provided. A test audio is played by a subwoofer and a main speaker, transmitted audio is collected by an audio collection device at a primary listening position, a sound generation delay and a volume parameter difference are analyzed, and the subwoofer is calibrated based on of a frequency response curve of a scenario, so as to solve the problem in the related art where the user experience is adversely affected due to low sound calibration accuracy between the subwoofer and the main speaker.

An augmented reality display system comprises a passable world model data comprises a set of map points corresponding to one or more objects of the real world. The augmented reality system also comprises a processor to communicate with one or more individual augmented reality display systems to pass a portion of the passable world model data to the one or more individual augmented reality display systems, wherein the piece of the passable world model data is passed based at least in part on respective locations corresponding to the one or more individual augmented reality display systems.

Audio mixed reality devices and systems that may enhance, diminish, or modify real-world sounds, such as by overlaying audio streams onto real-world sounds or by combining audio streams with real-world sounds. The audio mixed reality devices and systems may enable presence in the real world, immersion in audio streams, or focus on real-world sounds. Immersive spatial audio scenes may be layered over or otherwise used to modify real-world sounds. An audio mixed reality device may capture and reproduce real-world sounds such that they appear spatially situated in the user's environment. The resulting auditory experience may ensure that real-world sounds maintain their natural origin points, thereby providing an acoustic foundation upon which audio streams can be overlaid or modulated without disorienting the user. By dynamically reconstructing a spatial representation of the external environment, the audio mixed reality device may facilitate the contextual and spatial interaction of audio streams with real-world sounds.

The present disclosure generally relates to interfaces and techniques for media playback on one or more devices. In accordance with some embodiments, an electronic device includes a display, one or more processors, and memory. The electronic device receives user input and, in response to receiving the user input, displays, on the display, a multi-device interface that includes: one or more indicators associated with a plurality of available playback devices that are connected to the device and available to initiate playback of media from the device, and a media playback status of the plurality of available playback devices.

In one aspect, the systems and methods described herein may include a first device comprising a first ultra-wideband (UWB) antenna. The first device may establish a connection with a second device having a second UWB antenna. The first device may determine an orientation of the first device relative to the second device, according to one or more UWB measurements between the first UWB antenna of the first device and the second UWB antenna of the second device. The first device may receive audio signals corresponding to the second device, and can render the audio signals into audio output to a user of the first device according to the determined orientation.

A passenger conveyance system (100; 200; 300; 400; 500; 600) includes a directional speaker system (106; 206; 306; 406a-c; 506a-b; 606) configured to transmit an audio communication (120a-b; 220a-b; 320a-b; 420a-e) into a restricted target area (110a-b; 310a-b; 410a-b; 610a-c). Using a directional speaker system allows audio communications to be sent to a user (or to multiple users) without broadcasting the message more widely.

Audio / video base stations may include a plurality of audio / video input ports. The base stations may include an audio / video output port. The base stations may include a communication interface that is coupleable with an audio transceiver. The base stations may include one or more processors. The base stations may include a memory device having instructions stored thereon that, when executed by the one or more processors, cause the audio / video base station to detect that the audio transceiver has been undocked from the audio / video base station. The instructions may further cause the base station to, in response to detecting that the audio transceiver has been undocked from the audio / video base station, automatically switching an audio output signal from the audio / video output port to the communication interface while continuing to transmit a video output signal to the audio / video output port.

A method for rendering an audio source using a plurality of virtual sources is provided. The plurality of virtual sources includes a first virtual source. The method comprises obtaining a target distance gain value that was derived using a target distance gain function for the audio source and a reference distance value indicating a distance between a listening position and a reference point for the audio source. The method further comprises deriving a first distance gain correction value for at least the first virtual source using the target distance gain value. The method further comprises rendering the audio source using the derived first distance gain correction value and a signal for the first virtual source.

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1 ≤ m < n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

An audio output device (e.g., earbuds or headphones) includes one or more audio output components. While outputting, via the one or more audio output components, first audio content corresponding to a first set of one or more audio streams, the device detects an occurrence of an event involving a proximate audio source. In response to detecting the occurrence of the event, the device outputs, via the one or more audio output components, second audio content corresponding to a second set of one or more audio streams, the second set of one or more audio streams including an audio stream from the proximate audio source.

A system for controlling a plurality of lighting devices (11-15) to render light effects accompanying a rendering of audiovisual content (81) is configured to associate, based on a first characteristic of a first audio channel / object (86), the first audio channel / object with a first lighting device (12) and not with a second lighting device (13). The first characteristic is indicative of an audio source position. The system is further configured to associate a second audio channel (87) with the first and second lighting devices. The system is further configured to determine whether second audio content in the second audio channel meets predetermined criteria and determine a first light effect based on a determined chromaticity. If the one or more predetermined criteria are not met, the light intensity is based on first audio content in the first audio channel / object, and otherwise, the light intensity is based on the second audio content.