6939 results about "Audiology" patented technology

A system, including: an ear worn device configured to provide audio to a user's ear; a switch configured to receive physical contact from the user and, in response, alter an activation state of the switch; and a case configured to receive the ear worn device when not in use, wherein the case includes an electromagnet configured to magnetically attract the ear worn device, wherein changing the activation state of the switch causes the electromagnet to reduce attraction to the ear worn device to allow the user to more easily remove the ear worn device from the case.

Systems, methods, and computer program products are disclosed for establishing nonvocalized communications. Establishing nonvocalized communications may include detecting a directional gesture made by a wearer of a first wearable device configured to determine facial skin micromovements of the wearer, wherein the directional gesture identifies a second device in proximity to the first wearable device. Based on the directional gesture, a wireless communication channel for enabling a nonvocalized communication between the first wearable device and the second device is selected. Thereafter, specific facial skin micromovements of the wearer of the first wearable device are determined. The specific facial skin micromovements indicate words to be communicated in an absence of perceptible vocalization by the wearer of the first wearable device. Then, the words are transmitted via the wireless communication channel from the first wearable device to the second device for presentation via the second device.

A user electronic device comprising: one or more microphones configured to capture raw audio data; and one or more processors and one or more storage devices storing instructions that when executed by the one or more computers cause the one or more computers to perform operations comprising: receiving the raw audio data captured by the one or more microphones; processing the raw audio data using a speech transcriber to generate a live transcription of the raw audio data that comprises a plurality of text tokens; processing the raw audio data to generate a speaker identification output that identifies, for each of the text tokens, a respective speaker for each of the text tokens in the live transcription; and processing a first input comprising (i) a first input prompt and (ii) an input text generated from the live transcription using a language model neural network to generate a modified transcription.

A loudspeaker apparatus includes a circuit housing configured to accommodate a circuit component or a battery; an ear hook; a housing of an earphone core configured to accommodate the earphone core; and a housing protector at least partially covering a periphery of the circuit housing and the ear hook. A first end of the ear hook is connected to the circuit housing. The earphone core is driven by the circuit component or the battery to vibrate to generate sound. The housing of the earphone core is connected to a second end of the ear hook away from the circuit housing through a hinge component. The hinge component is capable of rotating to change a position of the housing of the earphone core relative to the ear hook.

The invention relates to the technical field of artificial intelligence, can be applied to the field of medical health, and discloses a timbre conversion method, device, equipment and medium based on cultural semanteme, the method comprises the following steps: constructing a cultural semantic timbre library comprising a semantic label and timbre characteristic parameter mapping relationship, the semantic label characterizing emotional semanteme of a target timbre, and the timbre characteristic parameter mapping relationship between the semantic label and the timbre characteristic parameter; the timbre characteristic parameters comprise a pitch range, rhythm rhythm and a harmonic structure; performing feature extraction based on text, image and audio multi-mode information to obtain semantic keywords, visual emotion features and audio acoustic features; performing attention weight fusion on the features through a multi-modal fusion deep learning model, and dynamically adjusting model parameters in combination with a semantic timbre library to generate a target timbre; and finally, intelligent conversion from the multi-mode information to the adaptive tone is realized. Through semantic-driven multi-modal feature collaborative optimization, the defect that timbre conversion machinery is stiff and lacks emotional expression is overcome, and the integrating degree of timbre expression and semantic scenes is improved.

The present disclosure relates to a speaker device. The speaker device may include an ear hook, a core housing for accommodating an earphone core, and a circuit housing for accommodating a control circuit or a battery. The ear hook may include a first plug end and a second plug end. The ear hook may be surrounded by a protective sleeve which may be made of an elastic waterproof material. The ear hook may be elastic, and a position of the core housing relative to the ear hook may be changed according to an elastic deformation of the ear hook, thereby the core housing may fit a user in front of or behind an ear of the user. The circuit housing may be fixed to the second plug end. The control circuit or the battery may drive the earphone core to vibrate to generate a sound.

The invention discloses a method for realizing synchronization of a character expression and a lip shape in a video through an emotion in sound and a cloned digital human system, and the method for realizing synchronization of the character expression and the lip shape in the video through the emotion in the sound comprises the steps: collecting audio information, and extracting multi-dimensional features of the sound; applying a pre-trained expression and lip shape generation model to generate corresponding expression parameters and lip shape parameters according to the multi-dimensional features of the sound; performing fusion processing on the expression parameters and the lip shape parameters, and generating a continuous animation sequence according to the fused parameters; and rendering the continuous animation sequence to generate video information with expressions, lip shapes and sound emotion synchronization. The high-precision expression and lip shape synchronization is realized, the natural fidelity of the cloned digital human is improved, and the application range and value of the cloned digital human are expanded.

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.

The invention relates to the technical field of electroencephalogram signal processing, in particular to an ontology brain wave audio auditory perception synchronous feedback method, device and system and electronic equipment. The method comprises the following steps: receiving electroencephalogram signals of a collected user on line from electroencephalogram collection equipment through an upper computer, obtaining electroencephalogram signals of a specified frequency band from the electroencephalogram signals, and extracting corresponding electroencephalogram characteristics; according to the electroencephalogram features or preset rhythm parameters, the electroencephalogram signals of the specified frequency band are segmented into a plurality of electroencephalogram segments, a plurality of audio expressions corresponding to the electroencephalogram segments are generated, and feature parameters of the audio expressions are determined according to the electroencephalogram features of the corresponding electroencephalogram segments; generating brain wave audio representation data according to the audio representation corresponding to the electroencephalogram signals of the one or more designated frequency bands, and obtaining brain wave audio according to the brain wave audio representation data. Therefore, the physiological suitability of nerve regulation and control and the artistic expressivity of audio generation are met at the same time, and organic unification of nerve regulation and control and audio generation is achieved.

This document relates to employing tongue gestures to control a computing device, and training machine learning models to detect tongue gestures. One example relates to a method or technique that can include receiving one or more motion signals from an inertial sensor. The method or technique can also include detecting a tongue gesture based at least on the one or more motion signals, and outputting the tongue gesture.

Systems, methods, and non-transitory computer-readable media containing instructions are disclosed for detecting and utilizing facial skin micro-motion. In some non-limiting embodiments, detection of facial skin micro-motion occurs using a speech detection system that may include a wearable housing, a light source (coherent light source or incoherent light source), a light detector, and at least one processor. The one or more processors may be configured to analyze light reflections received from the facial region to determine facial skin micromotion, and extract meanings from the determined facial skin micromotion. Examples of meanings that may be extracted from the determined facial skin micromotion may include words spoken by the individual (silent or voiced), an identity of the individual, an emotional state of the individual, a heart rate of the individual, a respiratory rate of the individual, or any other biometric, emotion or speech related indicator.

The invention relates to a sound effect adjustment method, a training method of a sound effect adjustment multi-modal large model, computer equipment and a storage medium. The method comprises the following steps: in response to a sound effect adjustment request for target music sent by a terminal, obtaining an original audio of the target music; inputting the original audio of the target music into a target encoder module of the trained sound effect adjustment multi-mode large model to obtain audio features of the target music, inputting the audio features into a projection module, and converting the audio features into music description semantic features of the target music; the music description semantic feature is a semantic feature of a description text of the target music; and inputting the music description semantic features of the target music into the large language model module to obtain sound effect adjustment parameters of the target music. By adopting the method, a user does not need to manually select a sound effect adjustment mode, and the adjusted parameters can be determined according to the target music, so that the sound effect adjustment effect can be improved.

The embodiment of the invention provides a method and a device for identifying the authenticity of sound in audio information, which can refine classification categories under the condition of identifying that the sound in the audio information is real sound or forged sound (such as synthetic sound), and specifically, the method and the device can be used for identifying the authenticity of the sound in the audio information. A real sound or counterfeit sound classification category is each refined into at least one hidden category within a hidden space, a single hidden category being characterized by a single prototype vector. According to the method, after the audio information to be recognized is coded to obtain the corresponding coding vector, the coding vector can be compared with each prototype vector to obtain each corresponding similarity, and then the sound authenticity of the audio information to be recognized is determined according to each similarity, namely, the audio information belongs to a real sound classification category or a forged sound classification category. Therefore, the accuracy of sound authenticity identification in the audio information can be improved.

The invention relates to the technical field of echo detection, in particular to an active sonar echo detection method, which comprises the steps of collecting active sonar reverberation data; constructing an active sonar channel, and generating reverberation data of a target echo; extracting target echo signal features by using matched filtering; a classification model of an improved SwinTransform is constructed, pooling, full connection and residual structures are constructed behind a Patch Merge layer of a hierarchical model of different stages, a transient target peak value in a reverberation background is highlighted, and time-frequency correlation characteristics of reverberation and a target are captured. According to the method, the problem that the accuracy of predicting whether a target exists or not by using sonar echoes through a Swin Transformer model needs to be further improved is solved.

An example wearable device for placement relative to an ear includes a housing; a first extending structure; a second extending structure, the first and second extending structures are separated by an adjustable distance; a carriage coupled to the second extending structure and being at least partially disposed within the housing, the carriage being configured to move longitudinally along the housing and an adjustment mechanism configured to impart a force to cause longitudinal movement of the carriage to adjust the adjustable distance.

Implementations described herein relate to detecting hallucinations in responses generated by large language models (LLMs). A natural language (NL) based input associated with a client device may be received. A first LLM response may be generated based on processing the NL based input using an LLM. Based on processing the first LLM response, it may be determined whether the first LLM response contains at least one hallucination. Responsive to determining that the first LLM response contains at least one hallucination, a second LLM response may be generated based on processing the NL based input. It may be determined whether the second LLM response contains at least one hallucination, and, responsive to determining that the second LLM response does not contain at least one hallucination, the second LLM response may be caused to be rendered at the client device.

A system for conducting voice modification on an audio input signal comprising speech to obtain an audio output signal according to an embodiment is provided. The system comprises a feature extractor for extracting feature information of the speech from the audio input signal. Moreover, the system comprises a fundamental frequencies generator to generate modified fundamental frequency information depending on the feature information, such that the modified fundamental frequency information comprises modified fundamental frequencies being different from real fundamental frequencies of the speech, and / or such that the modified fundamental frequency information indicates a modified fundamental frequency trajectory being different from a real fundamental frequency trajectory of the speech. Furthermore, the system comprises a synthesizer for generating the audio output signal depending on the modified fundamental frequency information and depending on the feature information.

In aspects of ending active noise cancellation based on a detected audio source, a mobile device implements an audio playback manager that detects a headset employing active noise cancellation (ANC) of audio in communication with a mobile device in an environment of the mobile device. The audio playback manager receives a selection of at least one audio source that triggers adjustment of the ANC and detects sound originating from the at least one audio source that triggers adjustment of the ANC. Based on detecting the sound originating from the at least one audio source, the audio playback manager adjusts the ANC.

The invention relates to the field of biological medicine, and discloses a method and system for analyzing cough sound by using disease characteristics to diagnose respiratory diseases, and the method comprises the steps: deploying a six-microphone annular array to achieve the precise positioning and triggering of a sound source; self-adaptive spectral subtraction and Wiener filtering cascade are adopted to enhance the audio; segmenting a cough segment based on energy envelope; fusing the Mel-cepstrum, the linear prediction residual error, the harmonic energy ratio and the transient zero-crossing rate to construct a pathological feature matrix; extracting local, medium-range and global time sequence features through a three-branch parallel convolutional network; inputting a disease specific classifier to discriminate asthma, pneumonia and laryngitis respectively, and applying a feature decoupling regular term to improve interpretability. The system correspondingly realizes the modularized processing flow. According to the method, the cough sound collection quality and the disease subtype recognition accuracy in a complex environment are improved, meanwhile, the thermodynamic diagram is output to assist clinical decision making, and the diagnosis credibility and practicability are enhanced.

A work machine includes a lower traveling body; an upper slewing body slewably mounted on the lower traveling body; an external microphone configured to collect sound around a work machine; an internal speaker configured to output sound to an operator of the work machine; an information transmission device configured to enable a worker around the work machine to recognize whether or not a voice of the worker has reached the operator of the work machine; and a processor, and a memory storing instructions that cause the processor to execute a process. The process includes collecting, by the external microphone, sound around the work machine; outputting, by the internal speaker, the sound to the operator of the work machine; and enabling, by the information transmission device, the worker around the work machine to recognize whether or not a voice of the worker has reached the operator of the work machine.

A body-worn PAP system providing proper mask fitment during body movement, reduced leakage, and elimination of a need for a connecting hose to the blower unit, and a PAP blower adapted for use in a PAP system with a compact noise reduction system that does not utilize acoustic foams in the airflow path.

Methods and apparatuses are described for modulating multiple integrated neural networks to alter composite sensory processes, such as audition or hearing. These methods and apparatuses may be used for therapeutic and non-therapeutic uses, including enhancing entertainment and communication, by providing a cranio-cervical tuning apparatus worn in or around the outer ear or auricle. These methods and apparatuses may include functional neurosensory bias for neurosensory scrambling neuromodulation to influence composite or multi-modal sensory processes.

A gesture recognition system for recognising gestures from input data and outputting gesture events may control a computing system. The gesture recognition system may include a plurality of devices, including one or more peripheral devices and a central device. Each peripheral device may include a set of one or more sensors for sensing one or more parameters of a set of parameters; a communication module configured to communicate with another device of the plurality of devices; and a processor. The processor is configured to process the sensor data using a rules engine and output a gesture event. The central device may include a central communication module for communicating with the one or more peripheral devices and a central processor configured, in dependence on processing one or both of (i) at least a subset of the sensor data and (ii) data relating to the gesture, to output a further gesture event.

A loudspeaker apparatus includes a circuit housing configured to accommodate a circuit component or a battery; an ear hook; a housing of an earphone core configured to accommodate the earphone core; and a housing protector at least partially covering a periphery of the circuit housing and the ear hook. A first end of the ear hook is connected to the circuit housing. The earphone core is driven by the circuit component or the battery to vibrate to generate sound. The housing of the earphone core is connected to a second end of the ear hook away from the circuit housing through a hinge component. The hinge component is capable of rotating to change a position of the housing of the earphone core relative to the ear hook.

The utility model relates to a novel empty drum. The novel empty drum comprises a drum body, the drum body comprises an upper drum body and a lower drum body which are fixedly connected, and eight peripheral tongues are arranged around the geometric center of the upper drum body; each peripheral tone tongue is provided with a first sub-tone tongue, a second sub-tone tongue, a third sub-tone tongue, a fourth sub-tone tongue and a fifth sub-tone tongue, and the first sub-tone tongue, the second sub-tone tongue, the third sub-tone tongue and the fourth sub-tone tongue are arranged close to the fifth sub-tone tongue; the geometric center of the lower drum body is provided with a timbre control hole and a rubber plug matched with the timbre control hole. According to the novel empty drum, the structure of the peripheral tongues is improved and designed, so that the tone of the empty drum in the playing process is richer, and the use experience of a user is improved.

The invention provides a head-mounted OWS open earphone. The head-mounted OWS open earphone comprises a head beam and two earphone bodies, the two earphone bodies are connected with the two sides of the head beam respectively, and at least one earphone body comprises an ear frame, a sound production unit and a rotary cover; the inner side of the ear rack is provided with an accommodating cavity for accommodating an ear, the sound production unit is mounted on one side of the ear rack, a part of structure of the sound production unit is located in the accommodating cavity, at least one air hole is formed between the sound production unit and the ear rack, and the air hole is communicated with the accommodating cavity; the rotary cover is rotatably mounted on the ear frame and opens or closes the air hole during rotation so as to switch the earphone mode; according to the earphone, the air hole can be opened or closed through rotation of the rotary cover so as to switch the earphone mode, so that the earphone provides two use modes, the two use modes can be switched randomly so as to adapt to requirements of different users and adapt to different environments, the earphone is very convenient and flexible to use, and the use experience can be effectively improved.

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.

An ultrasound transducer assembly is connectable to an ultrasound system and comprises one or more ultrasound transducer elements supported by a cap. The ultrasound transducer elements are operable to direct ultrasound energy toward brain tissue of a subject and / or to receive echo ultrasound energy when the ultrasound transducer assembly is mounted on the head of the subject. Some embodiments include a fillable jacket coupled to the inner surface of the cap and in acoustic contact with the one or more transducer elements.