Abstract: Partially adaptive audio beamforming systems and methods are provided that enable improved acoustic echo cancellation of sound played on a loudspeaker that is in close proximity to a microphone array in an audio device. A stored beamformer parameter, such as an inverse covariance matrix, can be utilized by a frequency domain beamformer to generate a beamformed signal. The overall performance and resource usage by the audio device can be optimized.

Abstract: A method for mixing a plurality of input signals, an apparatus and a computer program product are provided. The method comprises receiving a plurality of current power values associated to a current time interval and a plurality of previous smoothed power values associated to a previous time interval, when it is determined that at least one of the plurality of input signals contains speech, calculating the current smoothed power value for each input signal based on a current power value and a previous smoothed power value, when it is determined that none of the plurality of input signals contains speech, calculating the current smoothed power value for each input signal based on a determined value and the previous smoothed power value corresponding to each input signal and calculating a plurality of mixing gains based on the plurality of current smoothed power values.

Abstract: Techniques are described for maintaining contextual data to support content-related actions. In an example, a system stories second content at a source. The source is associated with first content. The system sends, to a device, an object that indicates the first content. From the device at a first time, the system receives first data indicating a first request for the second content and including source information that indicates the source. From the device at a second time, the system receives second data indicating a second request for the second content, the second data including the source information, the first data and the second data received at a frequency indicated by the object. The system determines that the requests are associated with the first content based on the source information included in the received data, and stores third data indicating a presentation of the first content by the device.

Abstract: An audio device comprising memory, an interface, and one or more processors, wherein the one or more processors are configured to obtain audio data; process the audio data for provision of an audio output; determine an uncertainty parameter based on the audio data; and control the processing of the audio data for provision of the audio output based on the uncertainty parameter.

Abstract: A system includes a first microphone that captures audio, a communication module communicatively coupled to the first microphone, a logic circuit communicatively coupled to the first microphone and communication module, a speaker operatively coupled to the logic circuit, and an interaction element. The interaction element and logic circuit are configured to initiate control of audio content for output from the speaker in response to at least one voice command detected in captured audio. Other embodiments are disclosed.

Abstract: A computer-implemented system is disclosed which enhances legal research by utilizing a first user interface (UI) to receive search criteria as text prompts from users, which are then processed by an application program interface (API) across an electronic connection. The system includes an instruction base for expanding the prompts into snippets for a machine learning network, which defines the legal database searches. The process further refines the prompts through a transformation module, breaking down and tagging the text for compatibility with the machine learning network. Coupled with non-transitory computer memory, the system efficiently outputs results in response to the applied search criteria. A second interactive UI displays these results and allows for user interaction to delve into detailed information, optimizing the legal information retrieval process. Also disclosed is a method for processing legal research queries enhanced by a machine learning network.

Abstract: Disclosed are a multi-speaker overlapping voice detection method and a system. The method includes: obtaining a voice to be detected, and removing silence from the voice to be detected is removed; extracting a feature of the voice to be detected after silence removal to obtain a voice feature of the voice to be detected; and inputting the voice feature into an overlapping voice detection model to obtain an overlapping speaker number corresponding to the voice to be detected output by the overlapping voice detection model. The overlapping voice detection model is obtained by supervised training based on a voice feature of a sample voice and a corresponding label of the overlapping speaker number, extracts an embedding of the voice feature, and classifies the overlapping speaker number to obtain the overlapping speaker number of the voice to be detected based on the extracted speaker embedding.

Abstract: A system on chip (SoC) for performing partial highlighting with the aid of auxiliary information detection includes a person recognition circuit, a sound detection circuit, an auxiliary information detection circuit and a processing circuit. The person recognition circuit obtains image data from an image capturing device, and performs person recognition on the image data to generate a recognition result. The sound detection circuit receives a plurality of sound signals from a plurality of microphones, and determines a voice characteristic value of a main sound. The auxiliary information detection circuit generates auxiliary information for calibrating the voice characteristic value of the main sound. The processing circuit determines a specific region in the image data according to the recognition result, the auxiliary information, and the voice characteristic value, and processes the image data to highlight the specific region.

Abstract: A method includes receiving location information of a user device; receiving a first and second microphone signals corresponding to an acoustic stimulus and a non-acoustic stimulus; processing the first and second microphone signals based on the location information of the user device; generating a first and second compensated signals and an average signal corresponding to an average of the first and second compensated signals; responsive to the determining that an instantaneous magnitude of a first signal is greater than that of a second signal, generating an output audio signal by switching or cross fading between a beamformed signal and an alternative signal such that a contribution of the alternative signal to the output audio signal is increased and a contribution of the beamformed signal to the output audio signal is decreased.

Abstract: Methods, systems, and apparatuses are provided for predicting if an item of text data includes one or more sensitive terms. An item of text data may be received by a computing device from a second computing device associated with a user. The computing device may determine one or more words within the item of text data. An arrangement of the one or more words may be determined. The arrangement of the one or more words may be based on an assignment bearing vectorization of the item of text data and/or a position bearing vectorization of the item of text data. Based on the arrangement of the one or more words within the item of text data, the computing device may determine a potential sensitive term within the item of text data. The computing device may cause an output of an indication of the potential sensitive term.

Abstract: 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).

Abstract: A device, system and method for AI-based noise suppression is provided. The device is configured to perform during a first period of time: applying one or more AI algorithms to an audio data; applying noise suppression to the audio data to generate a noise-suppressed audio data; and providing the noise-suppressed audio data to an output device. The device is further configured to perform during a second period of time, the second period of time following the first period of time: applying the one or more AI algorithms to the audio data to generate an AI-based noise-suppressed audio data; and providing the AI-based noise-suppressed audio data to the output device.

Abstract: The invention provides a method for identifying similarity between two audio files or tracks. The method comprises receiving a processed audio file and an original audio file, uncompressing the processed audio file, applying global loudness normalization and short-term loudness normalization on the processed audio file and the original audio file, converting the processed audio file and the original audio file into processed spectral image by time-frequency mapping, scaling, using linear interpolation, the processed spectral image, dividing the scaled-up processed spectral image into slices, searching for minimum Sum of Absolute Difference (SAD), using original spectral image as reference, for each slice.

Abstract: A high-quality voice signal processing device through removal of ambient noise based on multi-sensor signal fusion, includes: a voice microphone sensor that senses and outputs a speaker's voice signal; an accelerometer sensor that senses vibration of the speaker's vocal cords and outputs a signal; a noise reduction processing MCU that extracts a voice section according to vocal cord vibration using the output signal of the accelerometer sensor, synthesizes a low-frequency component of the accelerometer sensor and a low-frequency component of the voice microphone sensor at different synthesis ratios based on a level of noise extracted from the output signal of the voice microphone sensor using voice section information, and restores and outputs a voice signal by adding the synthesized low-frequency components and a high-frequency component of the voice microphone sensor; and a wireless communication module that externally outputs the restored voice signal.

Abstract: Disclosed are a directional bilateral sound intake-based MIC assembly and an electronic device. The electronic device includes the directional bilateral sound intake-based MIC assembly. The MIC assembly includes a front shell with paired sound inlet holes and a microphone arranged on an inner side of the front shell. The microphone has two receiving holes. The MIC assembly further includes a fixture assembled within the inner side of the front shell. An assembling chamber and two sound inlet channels in communication with the assembling chamber are defined in the fixture. The microphone is mounted within the assembling chamber. The two receiving holes of the microphone are in communication with the two sound inlet channels respectively. The two sound inlet channels are in communication with a same pair of two sound inlet holes respectively.

Abstract: A method comprising capturing, using a first capturing mode, immersive audio using a first capturing device comprising a first microphone and a second capturing device comprising a second microphone, recognizing, based on obtaining data from one or more sensors, movement of the first capturing device, wherein the movement is with respect to the second capturing device, recognizing the movement as movement for changing from the first capturing mode to a second capturing mode, wherein the second capturing mode is for capturing immersive audio, and capturing the immersive audio using the second capturing mode.

Abstract: Systems and methods are described for recognizing and responding to commands in a virtual or physical environment. A system may receive voice data and determine an intended command. The system may then determine a position and viewpoint orientation of the user to be able to determine one or more digital assets associated with the user. The system may then determine a current state associated with each digital asset of the one or more digital assets to be able to determine at least one digital asset that is configured to process the command. The system can then apply the command to at least a first digital asset of the at least one digital asset that is configured to process the command.

Abstract: A communication method and a communication assembly, comprising a first capturing unit configured to capture video data of a first peer in real-time, wherein the video data of the first peer comprises an identification feature a mimic expression feature; a second capturing unit configured to capture video data of a second peer in real-time; a first anonymization unit configured to modify the video data captured by the first capturing unit in real-time by replacing a characteristic of the identification feature in the video data of the first peer while substantially maintaining a characteristic of the mimic expression feature in the video data of the first peer; a first display unit of the first peer; a second display unit of the second peer; a data transfer unit configured to perform a transfer of the video data.

Abstract: An audio device comprising an interface, a memory, and one or more processors is disclosed, wherein the one or more processors are configured to: obtain a microphone input signal; obtain a media input signal; process the microphone input signal for provision of a microphone output signal; and provide an audio output signal based on the microphone output signal, wherein to process the microphone input signal comprises to: determine a microphone gain; and apply the microphone gain to the microphone input signal for provision of the microphone output signal, and wherein to determine the microphone gain comprises: estimate a first loudness of the microphone input signal; determine a first primary average based on the first loudness; estimate a second loudness of the media input signal; determine a second average based on the second loudness; determine a first gain based on the first primary average and the second average; determine a first secondary average based on the first loudness; determine a second gain bas

Abstract: A presentation service generates an audience interface for an electronic presentation. The audience interface may simulate an in-person presentation, including features such as a central presenter and seat locations for audience members. The audience members may select emotes which may be displayed in the audience interface. The emotes may indicate the audience members' opinion of the content being presented. The presentation service may enable chats between multiple audience members, grouping of audience members private rooms, and other virtual simulations of functions corresponding to in-person presentations.