7084results about "Stereophonic systems" patented technology

Some examples of the disclosure are directed to systems and methods for dynamically updating simulated source locations of audio sources within spatialized audio content based on detection of a change in the location and / or orientation of an audio output device (e.g., earbuds or speakers that are optionally worn by a user and are optionally included in a headset). In some examples, the simulated source locations are updated if the change in the location and / or orientation of the audio output device is greater than a threshold amount of change, and the simulated source locations are not updated if the change in the location and / or orientation of the audio output device is not greater than the threshold amount of change. In some examples, a simulated source location is identified in a three-dimensional environment. In some examples, a mode of outputting audio is changed in response to movement of an electronic device.

Some examples of the disclosure are directed to systems and methods for dynamically updating simulated source locations of audio sources within spatialized audio content based on detection of a change in the location and / or orientation of an audio output device (e.g., earbuds or speakers that are optionally worn by a user and are optionally included in a headset). In some examples, the simulated source locations are updated if the change in the location and / or orientation of the audio output device is greater than a threshold amount of change, and the simulated source locations are not updated if the change in the location and / or orientation of the audio output device is not greater than the threshold amount of change. In some examples, a simulated source location is identified in a three-dimensional environment. In some examples, a mode of outputting audio is changed in response to movement of an electronic device.

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).

Various implementations include audio devices and methods for spatializing audio output based on content. Certain implementations include at least one audio output device for providing an audio output based on data, and at least one controller coupled with the at least one audio output device, the controller configured to, use the data to determine a content type for the audio output from a group of content types, automatically select a spatialization mode for the audio output from a plurality of spatialization modes based on the determined content type, and apply the selected spatialization mode to the audio output.

An intelligent (self-learning) sensor-aware and / or context-aware subsystem comprising (i) a System-on-a-Chip (SoC), (ii) a radio transceiver, (iii) a microphone, (iv) a voice processing module, (v) a (bio-inspired) neuromorphic event camera or a hyperspectral camera, (vi) a first set of computer implemental instructions in artificial neural networks (ANN) (which may include a transformer model / diffusion model or Poisson flow generative model++ (PFGM++)), (vii) a second set of computer implementable instructions to analyze and interpret contextual data and (viii) an autonomous artificial intelligence (AI) agent is disclosed.

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.

The present disclosure generally relates to user interfaces for multi-user communication sessions. In some examples, a device initiates a live stream in a communication session. In some examples, a device transitions between streaming live audio and live video. In some examples, a device enables synchronizing media playback during a live stream. In some examples, a device displays synchronized media playback and plays a reaction from a first participant of the communication session.

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.

A device includes a memory configured to store audio data associated with an immersive audio environment. The device also includes one or more processors configured to obtain a listener pose in the immersive audio environment. The one or more processors are configured to determine whether an asset associated with the listener pose is stored locally at the memory. The one or more processors are configured to, based on the determination, select whether to retrieve the asset from the memory or to obtain the asset from a remote device. The one or more processors are also configured to generate an output audio signal based on the asset.

The invention provides an audio processing device and method, an augmented reality device, equipment and a storage medium, and belongs to the technical field of virtual reality display and sensing. The audio processing device comprises a positioning device which is configured to track and position a user in a real space in real time, determine positioning information of the user and send the positioning information to a processor; the processor is configured to determine target positioning information of the user based on pre-established spatial information and the real-time positioning information of the user; determining association relationship information between the user and a virtual sound source in a virtual space based on pre-established space information and real-time target positioning information of the user; obtaining human body parameters of the user, and determining an audio filtering result matched with the user based on the association relationship information, the human body parameters and the audio frequency of the virtual sound source; and reconstructing the audio to be played based on the audio configured for the virtual sound source in advance and the audio filtering result.

A system and method of targeting spatialized audio to ears of a listener, comprising sensing characteristics of an environment comprising at least one human by receiving radio frequency signals with a radio receiver; using the sensed characteristics in at least one automated processor to analyze at least one human dynamic physiological pattern of each human and estimate a body pose and head position within the environment of each human based on the sensed characteristics and the at least one human dynamic physiological pattern; and using an estimated position of the ears of each human in conjunction with a head-related transfer function in a spatial audio system to generate spatialized audio.

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).

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.

Approaches presented herein provide for identification of sound from a sound source relative to an array of microphones of a potentially unknown configuration using, in part, differences in the audio signals received by the microphones. In at least one embodiment, audio signals are captured using an array of microphones and audio features are extracted from those signals. The audio features can be processed using a first neural network to generate a feature vector representing a spatial location of an audio source with respect to the plurality of microphones, where the spatial location is inferred based on audio differences and independent of an availability of information indicating a physical configuration of the plurality of microphones. The feature vector can be provided to a task-specific model to perform at least one audio-related task based in part on the spatial location.

A method that includes determining a set of one or more reverberation parameters of an extended reality (XR) environment in which a user is to participate, determining whether an audio source device is wirelessly communicatively coupled to send audio to an audio output device, in response to determining that the audio source device is not wirelessly communicatively coupled to the audio output device obtaining a microphone signal produced by a microphone of the audio source device, producing a reverberant audio signal from the microphone signal according to the set of one or more reverberation parameters, and sending the reverberant audio signal to drive a speaker driver of the audio source device.

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 embodiment of the invention provides an audio processing method and device, electronic equipment and a storage medium. The method comprises the following steps: determining a target pulse response of a first audio signal, wherein the target pulse response is a stereo pulse response used for simulating sound reflection and attenuation effects of the first audio signal in an acoustic space; performing convolution processing on the first audio signal and the target pulse response to obtain a second audio signal; and performing audio mixing on the first audio signal and the second audio signal to output a third audio signal. According to the scheme provided by the invention, the stereo pulse response data capable of simulating the reflection and attenuation effects of the sound in the space can be synchronously determined when reverberation is carried out on the audio signal, and reverberation is carried out on the audio signal by using the stereo pulse response data, so that the sound is heard to be generated in different space environments; the reverberation effect of the audio signal is more vivid, the sense of space and the sense of layering of the sound are increased, and the sound is fuller and stereoscopic.

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.

A system for controlling one or more lighting devices (11-15) to render light effects while an audio rendering system (31, 34-39) renders audio content (81) is configured to obtain information indicative of audio rendering capabilities of the audio rendering system, obtain audio characteristics (83-88) of the audio content, select a subset of the audio characteristics based on the audio rendering capabilities of the audio rendering system, determine light effects based on the subset of the audio characteristics, and control the one or more lighting devices to render the light effects.

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.

An audio system is provided that includes a loudspeaker, at least one microphone, and at least one controller. The loudspeaker transmits an audio signal into a listening environment defined by at least one wall in a room. The at least one microphone is positioned on the loudspeaker and is configured to capture a reverberated audio signal including a plurality of reverberations and a plurality of peaks. The reverberated audio signal is indicative of the audio signal being reflected from the at least one wall. The at least one controller is programmed to apply a confidence score to the plurality of peaks to obtain a maximum score which is indicative of a maximum peak of the audio signal that is reflected from the at least one wall and determine a distance between the loudspeaker and the wall based at least on the maximum score.

A system for decoding Metadata Assisted Spatial Audio (MASA) signals in a pass-through coding mode 140 obtains an audio metadata signal 116 which encodes channel signals 102 and object information 104

The present invention provides a vehicle comprising a sound control device. A sound control device, according to one embodiment of the present invention, comprises: a sound object extraction module that extracts at least one audio object from an audio signal received from at least one of an outdoor microphone provided outside a vehicle and an indoor microphone provided inside the vehicle; and a sound object renderer that performs rendering with an audio function by using the audio object signal extracted by the sound object extraction module.

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.

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.

A method for providing navigation assistance using a head-worn device that has a camera, several microphones, and several speakers. The method captures sound in an environment as a plurality of microphone audio signals, captures, using the camera, a scene of the environment as a digital image, and processes the digital image to detect an object therein. The method selects, in response to the detection of the object, one of several navigation audio rendering modes. The several navigation audio rendering modes include a first mode that activates an acoustic transparency function to cause the speakers to reproduce the sound of the environment, a second mode that sonifies the object and activates the acoustic transparency function, and a third mode that sonifies the object, partially activates the acoustic transparency function, and an activates active noise cancellation function.

The document describes systems and techniques directed at three-dimensional, direction-dependent audio for multi-entity telecommunication. In aspects, a remote device receives multi-stream content, including at least one audio stream, from one or more audio-producing entities and obtains orientation information associated with the one or more audio-producing entities. The remote device can then, using the at least one audio stream and the orientation information, provide direction-dependent, three-dimensional audio sufficient to enable a multi-stereo audio output device to reproduce the spatial audio as if the at least one audio stream is originating from a direction, an elevation, and / or a proximity that corresponds to a physical location of the one or more audio-producing entities.

The invention discloses a self-adaptive sound effect management method and system with a scene analysis module, and relates to the field of self-adaptive sound effect management, and the method comprises the following steps: S1, collecting environmental acoustic characteristics and user behavior data in real time through the scene analysis module; s2, dynamically extracting user hearing preference parameters and scene classification labels; s3, constructing a virtual audition sweet zone matched with the physical space; and S4, dynamically adjusting the phase delay and the energy distribution proportion of each frequency band, and generating an optimal loudness equalization curve meeting the multi-scene constraint. According to the method, adaptive matching of sound field characteristics is realized in a complex dynamic scene through a multi-level perception-optimization closed-loop architecture, acoustic feature extraction and physical space modeling are deeply fused, the technical boundary of traditional pure signal processing is broken through, and the problem of multi-target conflict is solved based on cross-modal collaborative learning and a global optimization algorithm.