Abstract: To enable a patron of a venue attending a performance to feel as if he or she is ‘front and center’ even if located much farther away or angled from a stage of the venue, system and methods that enhance audio quality of audio signals especially over the voice frequency spectrum to generate a processed audio signal including an increased voice frequency spectrum are provided. The processed audio signal may be communicated to a receiving system with a headset or other transducer (e.g., loudspeakers) that allows for ambient sounds to be heard by a patron and for the patron to control volume of the processed audio signal, thereby mixing the processed audio signal with the ambient sounds. The receiving system may be rented and an operator may control usage for both performance or long-term rentals using various control features of the processed audio signals and/or receiving system.

Abstract: A system for facilitating configuration of an audio system tests the existing audio system configuration by playing reference audio clips of various different genres of programming, the audio of which is received by a microphone of a remote control device or other handheld mobile device at various different listening points in the room. The system then compares the audio signal representing audio of the reference audio clip received at the microphone to a reference audio signal representing the reference audio clip and determines other audio characteristics. The system then determines, based on the comparison and other characteristics of the room (e.g., furniture layout, room construction materials, wall treatments and current speaker positioning) suggested changes to a configuration of the audio system to increase audio quality of the audio system for playing audio associated with the various different genres of programming.

Abstract: Providing a more natural, physically accurate rendering of the acoustic behavior of a volumetric audio source (e.g., a line-like audio source). In one embodiment, this is achieved by applying a parametric distance-dependent gain function in the rendering process, where the shape of the parametric gain function depends on characteristics of the volumetric audio source.

Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: Device for reproducing sounds, in particular sounds intended to be perceived by a subject using said device, comprising a containment element in which a transducer, preferably of an electro-acoustic type, is housed, which is configured to emit vibrations substantially corresponding to the sound to be reproduced, and a liquid or gel state medium which is contained inside said containment element, said medium being configured to transmit the sound vibrations generated by said transducer to a transmission element intended to be applied in direct contact with a body area of said user, thereby transmitting said sound vibrations to said body area, wherein the containment element consists of a single housing chamber and the transducer is immersed entirely or partially in the medium inside said chamber.

Abstract: A speaker device is provided. The speaker device includes a frame, and a vibration unit, a magnetic circuit unit, and air-permeable isolation member. The magnetic circuit unit includes a yoke fixed to the frame. The yoke and the frame are spaced to form a leakage port to enable leaking from a side surface of the speaker device. The sound-producing inner cavity communicates with the leakage port. The air-permeable isolation member completely covers the leakage port. The leakage port communicates with the outside through the leakage hole. Compared with the related art, the speaker device of the present disclosure has a yoke leakage structure, which can realize self-sealing and is conducive to assembling.

Abstract: Disclosed is an audio signal processing method implemented by an audio system with internal and external sensors. The internal sensor measures acoustic signals propogating internally to a user's head. The external sensor measures acoustic signals propagating externally to the user's head. The method includes: producing first and second audio signals by measuring simultaneously acoustic signals reaching the internal and external sensors, respectively; filtering the second audio signal by a noise matching filter matching a second noise signal affecting the second audio signal with a first noise signal affecting the first audio signal, wherein the first noise signal and the second noise signal correspond to a same noise acoustic signal originating outside the user's head and measured by respectively the internal and external sensors, thereby producing a filtered second audio signal including a matched second noise signal; and mixing the filtered second audio signal and the first audio signal.

Abstract: A computing device is configured to perform functions comprising: receiving via a network microphone device of a media playback system, a voice command detected by at least one microphone of the network microphone device, wherein the media playback system comprises a plurality of zones, and the network microphone device may be a member of a default playback zone. The computing device may be further configured to perform functions comprising: dynamically selecting an audio response zone from the plurality of zones to play an audio response to the voice input and foregoing selection of the default playback zone. The selected zone may comprise a playback device, and the dynamically selecting may comprise determining that the network microphone device is paired with the playback device. The computing device may cause the playback device of the selected zone to play the audio response.

Abstract: An electronic device can include a housing defining an aperture and a display positioned in the aperture. The display and the housing can define an internal volume in which a speaker assembly is positioned. The speaker assembly can include a speaker module and a speaker enclosure in fluid communication, with the speaker enclosure at least partially defining a speaker volume.

Abstract: Apparatus for generating an enhanced sound field description, including: a sound field generator for generating at least one sound field description indicating a sound field with respect to at least one reference location; and a meta data generator for generating meta data relating to spatial information of the sound field, wherein the at least one sound field description and the meta data constitute the enhanced sound field description. The at least one sound field description includes in a certain case a first sound field description related to the at least one reference location and a second sound field description related to a further reference location being different from the reference location, and the meta data relating to the spatial information indicates the reference location and the further reference location or one of both reference locations and a vector extending between both reference locations.

Abstract: A method for audio beam steering, tracking, and audio effects for an immersive reality application is provided. The method includes receiving, from an immersive reality application, a first audio waveform from a first acoustic source to provide to a user of a headset, identifying a perceived direction for the first acoustic source relative to the headset based on a location of the first acoustic source, and providing, to a first speaker in a client device, an audio signal including the first audio waveform, wherein the audio signal includes a time delay and an amplitude of the first audio waveform based on the perceived direction. A non-transitory, computer-readable medium storing instructions which, when executed by a processor, cause a system to perform the above method, and the system, are also provided.

Abstract: Disclosed herein is a system for automatic detection of music listening reactions, comprising a wearable sensor and a mobile device. The wearable sensor is worn on the listener's ear. The mobile device receives an inertial signal and a sound signal of the listener from the wearable sensor. The mobile device determines a vocal reaction of the listener based on the inertial signal, the sound signal and music information of the music being played and a motion reaction of the listener based on the inertial signal and the music information. Other embodiments are described and shown.

Abstract: A system and method includes generating a virtual n-dimensional space that includes one or more positions of one or more source nodes and a position of a receiver node; executing a plurality of simulations including simulating acoustic signals emanating from the one or more source nodes within the virtual n-dimensional room; estimating a measure of the acoustic signals received at the receiver node; computing a plurality of acoustic signal data samples based on the estimation for each of the plurality of simulations; and creating a training data corpus for training an artificial neural network, the training data corpus including at least a sampling of the plurality of acoustic data samples, and the artificial neural network, once trained, is configured to generate an inference indicating a likely intended sound to a target receiver of a mixture of acoustic signals.

Abstract: The disclosure relates to a method for recognizing at least one naturally emitted sound produced by a real-life sound source in an environment comprising at least one artificial sound source. The method is implemented by an audio recognition device, and it includes simultaneously obtaining a first audio signal from a first microphone located in the environment and a second audio signal from an audio acquisition device associated with the at least one artificial sound source; analyzing the first audio signal, delivering a first list of sound classes corresponding to sounds recognized in the first audio signal; analyzing the second audio signal, delivering a second list of sound classes corresponding to sounds recognized in the second audio signal; and delivering a third list of sound classes, comprising only sound classes included in the first list of sound classes which are not included in the second list of sound classes.

Abstract: The present invention relates to a microphone module structure for an artificial intelligence smart device. According to the present invention, internally/externally generated noises, vibrations, impacts, or sounds output from a speaker are prevented from being transmitted to a microphone, whereby the voice recognition rate of an AI speaker, an AI soundbar, or various voice recognition devices can be significantly improved, and thus various smart devices can be effectively controlled via voice.

Abstract: A feeder/interconnection/return (FIR) Cable Loop System vastly improves the distribution, equalization, efficiency and power delivery of high performance multiple speaker systems, without the complexities and inherent inefficiencies of existing systems. This is achieved by a feeder cable which connects from an amplifier to the first speaker, while interconnecting cables connect all the speakers to each other. A return cable connects the last speaker back to the amplifier. This results in balanced power, significant improvement in frequency response, and dynamic range.

Abstract: A signal processing apparatus according an embodiment includes an acquisition unit and an application unit. The acquisition unit acquires M detection signals output from M detector devices having N-fold symmetry (M is an integer equal to or greater than 2, and N is an integer equal to or greater than 2). Each of the M detector devices detects original signals generated from K signal sources (K is an integer equal to or greater than 2) having the N-fold symmetry. The application unit applies a trained neural network to M input vectors corresponding to the M detection signals and outputs K output vectors. The same parameter is set to, of multiple weights included in a weight matrix of the trained neural network, weights that are commutative based on the N-fold symmetry.

Abstract: The present invention relates to a haptic motor driving method including: analyzing a received audio signal; extracting a corresponding haptic event signal when a result of the analysis is equal to or larger than a specific threshold value; generating a haptic signal corresponding to the extracted haptic event signal; and multiplexing pre-stored haptic data and the generated haptic signal by considering whether a first user input signal is received, and outputting a driving signal driving a haptic motor, and basically, a haptic signal is generated from an audio sound, but a system is configured in the form of responding to an input of a user, thereby providing a more realistic haptic result.

Abstract: A system and method for quieting unwanted sound. As a non-limiting example, various aspects of this disclosure provide a system and method, for example implemented in a premises-based or home audio system, for quieting unwanted sound at a particular location.

Abstract: Various implementations include audio systems and methods for mixed rendering to enhance audio output. Certain implementations include an audio system having: at least one far-field speaker configured to output a first portion of an audio signal; and a pair of non-occluding near-field speakers configured to output a second portion of the audio signal in synchrony with the output of the first portion of the audio signal, where the first portion of the audio signal is distinct from the second portion of the audio signal.