Abstract: Media input audio data corresponding to a media stream and microphone input audio data from at least one microphone may be received. A first level of at least one of a plurality of frequency bands of the media input audio data, as well as a second level of at least one of a plurality of frequency bands of the microphone input audio data, may be determined. Media output audio data and microphone output audio data may be produced by adjusting levels of one or more of the first and second plurality of frequency bands based on the perceived loudness of the microphone input audio data, of the microphone output audio data, of the media output audio data and the media input audio data. One or more processes may be modified upon receipt of a mode-switching indication.

Abstract: A wearable electronic device including a speaker and a microphone is configured to: generate a detection signal to detect whether a sound leakage occurs, receive, through the microphone, a feedback signal in which a signal obtained by outputting the detection signal into an ear canal of a user through the speaker is collected, calculate a difference value between the feedback signal and the detection signal, and correct a playback sound output by the wearable electronic device based on the difference value.

Abstract: Example techniques relate to voice interaction in an environment with a media playback system that is playing back audio content. In an example implementation, while playing back first audio in a given environment at a given loudness: a playback device (a) detects that an event is anticipated in the given environment, the event involving playback of second audio and (b) determines a loudness of background noise in the given environment, the background noise comprising ambient noise in the given environment. The playback device ducks the first audio in proportion to a difference between the given loudness of the first audio and the determined loudness of the background noise and plays back the ducked first audio concurrently with the second audio.

Abstract: In an embodiment, a system on a chip (SOC) includes a component that remains powered when a central processing unit (CPU) processor and a memory controller of the SOC are powered off. The component may include a sensor capture unit to capture audio samples from an audio detector circuit and write them to a memory of the component. A processor of the component may be configured to search the audio samples for a predetermined pattern during a time when the CPU processor and the memory controller are powered down. In some embodiments, based on the audio samples filling to a threshold level in the memory of the component and a lack of detection of the predetermined pattern, the component is configured to wake up the memory controller and a path to the memory controller in order to write the audio sample to a memory controlled by the memory controller.

Abstract: A wearable audio device includes a microphone, a speaker, and a hardware processor. The microphone is configured to receive external audio. The speaker configured to produce internal audio. The hardware processor is operably connected to the microphone and speaker. A comparison of the external audio to the internal audio is performed, and a notification is generated based upon the comparison determining that the external audio matches the internal audio.

Abstract: The present application relates to the field of signal processing, and in particular, to a hearing aid method and apparatus for noise reduction. A hearing aid method for noise reduction, comprising: identifying a scenario where a user is located; and if detection data contains sample data in a sample database corresponding to the scenario, entering a hearing aid mode, and in the hearing aid mode, playing back all or part of external sounds, the external sounds being acquired by a reference microphone; the sample database is a sample sound library, and the sample data is a sample sound; different scenarios correspond to different sample sound libraries; wherein the sample sounds in respective sample sound libraries are configured with priorities; and the sample sounds in the sample sound libraries corresponding to different scenarios have different priorities.

Abstract: A system comprising audio processing circuitry is provided. The audio processing circuitry is operable to receive audio signals. The audio processing circuitry is operable to process the audio signals to detect strength of a chat component of the audio signals and strength of a game component of the audio signals. The audio processing circuitry is operable to automatically control a volume setting based on one or both of: the detected strength of the chat component, and the detected strength of the game component. The combined-game-and-chat audio signals may comprise a left channel signal and a right channel signal. The processing of the combined-game-and-chat audio signals may comprise measuring strength of a vocal-band signal component that is common to the left channel signal and the right channel signal.

Abstract: There is provided a voice output apparatus for providing a high-quality sound to an eardrum of a user. The voice output apparatus includes a first voice output unit outputting a voice to an ear canal of a user based on an output voice signal, a first noise acquirer arranged to face outward from a body of the user and captures a mixed voice including first external noise arriving from an outside of the user to output a mixed voice signal, an echo canceler cancelling an influence, on the first external noise, of a leaked voice output from the first voice output unit and leaking to the outside of the user, and a noise canceler generating a first external noise signal corresponding to the first external noise, and processing, using the first external noise signal, an input voice signal input from the outside to generate the output voice signal.

Abstract: A method for adjusting the clarity of an audio output in a changing environment, including: receiving a content signal; applying a customized gain to the content signal; and outputting the content signal with the customized gain to at least one speaker for transduction to an acoustic signal, wherein the customized gain is applied on a per frequency bin basis such that frequencies of a lesser magnitude are enhanced with respect to frequencies of a greater magnitude and an intelligibility of the acoustic signal is set approximately at a desired level, wherein the customized gain is determined according to at least one of a gain applied to the content signal, a bandwidth of the content signal, and a content type encoded by the content signal.

Abstract: Systems and methods for determining parameter adjustments for a capture of audio are disclosed. The systems and methods includes processing circuitry configured to access at least one energy map that corresponds to one or more audio streams. The processing circuitry may then determine, from the at least one energy map, a parameter adjustment with respect to at least one audio element. The parameter adjustment may be configured to adjust the capture of audio by the at least one audio element. In addition, the process circuitry may be configured to output an indication indicating the parameter adjustment with respect to the at least one audio element.

Abstract: A headset that can detect the voice activity of a user includes an inner microphone generating an inner microphone signal; an outer microphone generating an outer microphone signal, wherein the inner microphone and outer microphone are positioned such that, when the headset is worn by a user, the inner microphone is disposed nearer to the user's head; and a voice-activity detector determining a sign of a phase difference between the inner microphone signal and the outer microphone signal and generating a voice activity detection signal representing a user's voice activity when the sign of the phase difference indicates that the outer microphone received an audio signal after the inner microphone received the audio signal.

Abstract: Methods, systems, computer-readable media, and apparatuses for audio signal processing are presented. A device for audio signal processing includes a memory configured to store instructions and a processor configured to execute the instructions. When executed, the instructions cause the processor to receive an external microphone signal from a first microphone and produce a hear-through component that is based on the external microphone signal and hearing compensation data. The hearing compensation data is based on an audiogram of a particular user. The instructions, when executed, further cause the processor to cause a loudspeaker to produce an audio output signal based on the hear-through component.

Abstract: In an example implementation, a method may involve, while a first zone and a second zone of a media playback system are playing back respective media, receiving data indicating the occurrence of a first trigger condition. The method may also involve, based on the received data, modifying respective volume limits of the first zone and the second zone, wherein modifying the volume limit causes first volume levels that exceed the second limit to be reduced to respective second volume levels that are at or below the second limit. The method may also involve receiving data indicating the occurrence of a second trigger condition. The method may further involve, based on the received data, modifying the respective volume limits of the first zone and the second zone from the second limit to the first limit.

Abstract: In some embodiments, a system comprises a host computing device and an audio device including at least one speaker and a plurality of microphones, the audio device being wirelessly and communicatively coupled to the host computing device. The host computing device can include one or more processors and one or more machine-readable, non-transitory storage mediums that include instructions configured to cause the one or more processors of the host computing device to perform operations including: receiving user environment data by one or more sensors of the host computing device; determining a characterization profile of a surrounding environment of the user based on the user environment data; and sending the characterization profile to the audio device, the characterization profile configured to cause the audio device to adapt an audio cardioid pattern of the plurality of microphones on the audio device based on the characterization profile.

Abstract: Systems and method are disclosed for facilitating efficient calibration of filters for correcting room and/or speaker-based distortion and/or binaural imbalances in audio reproduction, and/or for producing three-dimensional sound in stereo system environments. According to some embodiments, using a portable device such as a smartphone or tablet, a user can calibrate speakers by initiating playback of a test signal, detecting playback of the test signal with the portable device's microphone, and repeating this process for a number of speakers and/or device positions (e.g., next to each of the user's ears). A comparison can be made between the test signal and the detected signal, and this can be used to more precisely calibrate rendering of future signals by the speakers.

Abstract: The present application relates to the field of audio technology, and provides a method, a device, and an apparatus for playing audio, and a computer-readable storage medium. The method for playing audio includes: obtaining an ambient atmospheric pressure value and audio data to be played; obtaining multiple target frequency points contained in the audio data to be played when the ambient atmospheric pressure value meets a preset condition, and determining equal-loudness multiples corresponding to the target frequency points according to the ambient atmospheric pressure value and a preset calibration atmospheric pressure value; and sending the audio data to be played and the equal-loudness multiples of the target frequency points to a power amplifying module, such that the power amplifying module amplifies the audio data to be played according to the equal-loudness multiples corresponding to the target frequency points.

Abstract: Aspects of the present disclosure provide methods, apparatuses, and systems for closed-loop sleep protection and/or sleep regulation. According to an aspect, sleep disturbing noises are predicted and a biosignal parameter is measured to dynamically mask predicted disturbing environmental noises in the sleeping environment with active attenuation. Environmental noises in a sleeping environment of a subject are detected, input, or predicted based on historical data of the sleeping environment collected over a period of time. The biosignal parameter is used to determine sleep physiology of a subject. Based on the environmental noises in the sleeping environment and the determined sleep physiology, the noises are predicted to be disturbing or non-disturbing noises. For predicted disturbing noises, one or more actions are taken to regulate sleep and avoid sleep disruption by using sound masking prior to or concurrently with the occurrence of the predicted disturbing noises.

Abstract: Described herein is an audio device with a microphone which may adapt the audio output volume of a speaker by either increasing or decreasing output volume based on an audio input volume from a user and a distance from the user to the audio device. The audio device may also adapt its output volume to lower the audio output based on detecting one or more interruptions including occupancy and acoustic sounds.

Abstract: According to various embodiments, an electronic device comprises, a memory for storing instructions, and a processor for executing the stored instructions, wherein the processor is further configured, as the instructions are executed, to identify a trigger to execute a first event associated with a sound output of an external electronic device, to identify the location of a user within a space where the electronic device is located, to identify information about the sound at the location of the user, and on the basis of the information about the identified sound, to control the properties of a sound outputted by at least one of the external electronic device performing the first event or another external electronic device generating a sound.

Abstract: A bone anchored hearing device includes an electromagnetic vibrator for generating a vibration in order to transmit sound through a bone of a user to an ear of the user; and a compensator for at least in part compensating a distortion in the vibration of the electromagnetic vibrator. Further, a signal processing method for a bone anchored hearing device includes providing, by an input transducer, an electric input signal representing sound of a surrounding of a user of the bone anchored hearing device; processing, by a signal processing unit, the electric input signal and providing a processed electric signal; generating, by an electromagnetic vibrator, based on the processed electric signal, a vibration in order to transmit sound through a bone of the user to an ear of the user; and at least in part compensating, by a compensator, a distortion in the vibration of the electromagnetic vibrator.

Abstract: A vehicle includes: a seat; a sensor part provided on the seat and detecting whether a user is seated on the seat; a lighting part provided in the vehicle and including a plurality of light sources disposed at a preset interval; and a controller configured to determine a gaze direction of the user based on a result of the detection of the sensor part and to selectively control the plurality of light sources to distort the preset interval based on the determined gaze direction of the user.

Abstract: An electronic device includes: a memory storing instructions; and at least one processor configured to execute the instructions stored in the memory to control the at least one processor to: identify a location of the electronic device; obtain an image and a sound signal corresponding to the location; identify, using a trained neural network, a scene where the electronic device is present, based on the image, the sound signal, and the location; and provide settings of the electronic device based on the identified scene.

Abstract: The present invention provides a deep learning based method and system for processing sound quality characteristics. The method comprises: obtaining data characteristics of an audio data to be processed by extracting features from user preference data including the audio data to be processed; based on the data characteristics, generating a sound quality processing result of the audio to be processed by using a trained baseline model; wherein the baseline model is a neural network model trained by using audio data behavioral data, and other relevant data from multiple users or a single user.

Abstract: Example techniques may involve calibration with multiple recording devices. An implementation may include a mobile device receiving data indicating that a calibration sequence for multiple playback devices has been initiated in a venue. The mobile device displays a prompt to include the first mobile device in the calibration sequence for the multiple playback devices and a particular selectable control that, when selected, includes the first mobile device in the calibration sequence. During the calibration sequence, the mobile device records calibration audio as played back by the multiple playback devices and transmits data representing the recorded calibration audio to a computing device.

Abstract: An audio system includes a sound output device, a microphone, and processing circuitry. The microphone is configured to capture environmental audio. The processing circuitry is configured to analyze the environmental audio to identify one or more properties of environmental audio conditions. The processing circuitry is configured to adjust one or more sound presentation parameters based on the one or more properties of the environmental audio conditions to account for the environmental audio conditions. The processing circuitry is configured to operate the sound output device to output audio according to the one or more sound presentation parameters.

Abstract: A volume adjustment device includes a microphone, a speaker, and a signal processing unit. The at least one signal processing unit includes at least one processor and at least one memory. The signal processing unit is configured to output a measurement sound from the speaker. The signal processing unit is also configured to measure a first impulse response with the microphone. The signal processing unit is also configured to adjust a volume of the measurement sound output from the speaker based on a direct sound component of the first impulse response and an indirect sound component of the first impulse response.

Abstract: In some embodiments, a system comprises a host computing device and an audio device including at least one speaker and a plurality of microphones, the audio device being wirelessly and communicatively coupled to the host computing device. The host computing device can include one or more processors and one or more machine-readable, non-transitory storage mediums that include instructions configured to cause the one or more processors of the host computing device to perform operations including: receiving user environment data by one or more sensors of the host computing device; determining a characterization profile of a surrounding environment of the user based on the user environment data; and sending the characterization profile to the audio device, the characterization profile configured to cause the audio device to adapt an audio cardioid pattern of the plurality of microphones on the audio device based on the characterization profile.

Abstract: The concepts and technologies disclosed herein are directed to venue seat assignment based upon hearing profiles. According to one aspect of the concepts and technologies disclosed herein, a device can include a processor and a memory. The device can receive a request to upload a hearing profile, and can upload the hearing profile to a seat assignment system. The device can receive, from the seat assignment system, a customized seating chart based, at least in part, upon the hearing profile. The customized seating chart can include a visual representation of at least a portion of seating in a venue. The device can select, from the customized seating chart, a seat from the portion of the seating in the venue. This selection can be made automatically or based upon user input.

Abstract: An electronic apparatus and a method for controlling thereof are provided. The electronic apparatus includes a microphone configured to receive a user voice, a communication interface, and a processor configured, based on a first voice being received through the microphone, to provide first response information corresponding to the first voice, and based on a user sensing signal being received from an external apparatus through the communication interface, to control the communication interface to send the first response information to the external apparatus.

Abstract: A method performed by a wearable audio output device worn by a user is provided for controlling external noise attenuated by wearable audio output device. A speech is detected from a user wearing the wearable audio output device, wherein the audio output device has active noise reduction turned on. It is determined, based on the detecting, that the user desires to speak to a subject in the vicinity of the user. In response to the determining, a level of noise reduction is reduced to enable the user to hear sounds external to the audio output device. It is determined that the user desires to speak to the subject by detecting at least one condition of a plurality of conditions.

Abstract: A hearing device, e.g. a hearing aid, adapted for being located at or in an ear, or to be fully or partially implanted in the head, of a user, the hearing device comprises a) an input unit providing at least two input audio data streams, each comprising a mixture of a target signal component from a target sound source and a noise component from one or more noise sources; b) a mixing processor for receiving said at least two input audio data streams, and for mixing said at least two input audio data streams, or processed versions thereof, and for providing a processed input signal based thereon; c) an output unit providing output stimuli perceivable to the user as sound based on said processed input signal or a processed version thereof. The processor is configured to process said noise component of said at least two input audio data streams, or processed versions thereof in order to reduce or avoid artefacts in said processed input signal due to said mixing.

Abstract: Disclosed is a method and apparatus for determining one or more operation parameters for a dynamic range compression (DRC) system. The method comprises obtaining, as an input, a parameter indicative of a hearing ability of a user, the parameter relating to a first difference in sound intensity between a maskee at a first frequency and a masker at a second frequency, determining a target value for the parameter, and determining the one or more operation parameters such that a second difference in sound intensity after sound intensity modification by the DRC (between sound intensity of the maskee of the masker) corresponds to the target value for the parameter. The operation parameters are determined such that a dependence of the second difference in sound intensity on the sound intensity of the maskee is minimized for a given range of sound intensities of the maskee.

Abstract: Systems and methods for providing improved localization of recorded and played back sound are provided by improved microphone arrays for recording sound and by improved systems for playback of sound. Microphone arrays include four microphones with sound transducers located and aimed to mimic capture of sound by human ears. Sound captured by two side-viewing microphones is attenuated, at the time of sound capture and/or recording, at a later processing stage, or at the time of sound playback, by low-pass filtering. The recording maintains four separate channels of sound. Playback occurs through four speakers arranged to reproduce sound in the way human ears hear sound, with appropriate attenuation for side speakers. Playback can also occur through four-channel headphones. Improved playback of two-channel stereo sound can also occur through low-pass filtering of each track and playing the filtered sound through side/rear speakers on the opposite sides.

Abstract: According to one embodiment, an acoustic control apparatus includes a first calculator, a second calculator, and a first setting unit. The first calculator calculates a first relationship established between acoustic filter coefficients, based on sounds emitted from sound sources. The second calculator calculates a second relationship established between the acoustic filter coefficients by matching a first sound pressure ratio with a second sound pressure ratio, in a complex sound pressure ratio between ears of a user who desires the sound information. The first setting unit sets an acoustic filter coefficient corresponding to each of the sound sources, based on the first relationship and the second relationship.

Abstract: A method, system and computer-usable medium are disclosed for voice-based alerting of an individual wearing an obstructive listening device. Certain embodiments include detecting speech in an ambient environment in which the person wearing the obstructive listening device is present; determining whether the detected speech includes a name of the person wearing the obstructive listening device; if the detected speech includes the name of the person wearing the obstructive listening device determining whether the name of the person was spoken using a calling speech characteristic; and if the name of the person was spoken using a calling speech characteristic, automatically alerting the person wearing the obstructive listening device that another person in the ambient environment is calling for attention of the person wearing the obstructive listening device.

Abstract: To enable listening to audio in a more suitable mode without any complicated operation even in a situation in which the user's state or situation successively changes. An information processing device including: a recognition processing unit that recognizes a user's state in accordance with a detection result of a predetermined state or situation; and an output control unit that controls audio output from a predetermined output unit on the basis of a function map, which is selected in accordance with a predetermined condition, in which a setting related to control of the audio output is associated with each of a plurality of candidates for the user's state, and the recognized user's state.

Abstract: An electronic device and method that automatically adjusts an audio output volume level based on a live environmental acoustic scenario input via a microphone using a machine learning algorithm trained with Human Activity Recognition (HAR). Equipped with such an intelligence the electronic device classifies ambient sounds occurring in the environment of the listening area in which the device is situated into different acoustic scenario mappings such a voice or conversation, for an ambient human conversation detected event, and noise, such as for example a vacuum cleaner or dish washer noise detected event, and automatically adjust the audio output volume accordingly.

Abstract: Described herein is an audio device with a microphone which may adapt the audio output volume of a speaker by either increasing or decreasing output volume based on an audio input volume from a user and a distance from the user to the audio device. The audio device may also adapt its output volume to lower the audio output based on detecting one or more interruptions including occupancy and acoustic sounds.

Abstract: A method for spatial audio signal processing including: determining at least one first direction parameter for at least one frequency band based on microphone signals received from a first microphone array; determining at least one second direction parameter for the at least one frequency band based on at least one microphone signal received from at least one second microphone, wherein microphones from the first microphone array and the at least one second microphone are spatially separated from each other; processing the determined at least one first direction parameter and the at least one second direction parameter to determine at least one distance parameter for the at least one frequency band; and enabling an output and/or store of the at least one distance parameter, at least one audio signal, and the at least one first direction parameter.

Abstract: An input method for an electronic device comprises steps of retrieving operation data according to a touch point made by a user on an input interface; generating a pending input pattern in response to the operation data of the touch point; retrieving a matched character corresponding to the pending input pattern, in response to a defined mapping relation of a plurality of input patterns and a plurality of characters; outputting the matched character.

Abstract: Disclosed is an electronic device. A disclosed method for controlling an electronic device may include: generating a dedicated audio output descriptor and a dedicated audio output thread corresponding to an application in response to a preset event associated with the application, applying an audio setting value stored in the dedicated audio output descriptor to audio data associated with the application using the dedicated audio output thread, and mixing and outputting the audio data output using the dedicated audio output thread with audio data output from another audio output thread.

Abstract: A system for evaluating an ear seal between an earphone of a hearing device and an ear canal includes a first transducer that plays sound in response to an electrical signal that includes a reference frequency component and a test frequency component lower than the reference frequency. A second transducer receives the sound in the ear canal. A controller is configured to: calculate at least one electrical signal level difference between the electrical signal reference and test frequency components, measure acoustical levels of the reference and test frequency components of the sound in the ear canal, calculate an acoustical signal level difference between the measured acoustical levels of the reference and test frequency components, calculate a normalized acoustical difference value by subtracting the electrical signal level difference from the acoustical signal level difference, and determine a measurement of the ear seal based on the normalized acoustical difference value.

Abstract: The embodiment of the disclosure provides a volume adjusting method and device, a mobile terminal and a storage medium. The method includes: acquiring first audio information corresponding to an audio source when detecting that the audio source of a terminal device is playing; determining a first audio amplitude of the first audio information; determining a second audio amplitude of second audio information when the first audio amplitude meets a preset query condition, wherein the second audio information is played after the first audio information; determining corresponding adjustment information when the second audio amplitude meets a preset adjustment condition; and adjusting a volume of the terminal device according to the adjustment information. The disclosure improves the volume adjustment efficiency and reduces the energy consumption of the terminal device.

Abstract: A device is used to monitor, in real-time, one or more environment conditions of an environment in which a user is located. Based on the monitoring, an alert condition relating to the environment is detected. Based on detecting the alert condition, on-demand testing of a sensory component of the user is initiated. The on-demand testing tests for a selected condition relating to the health of the user.

Abstract: For adjusting speaker volume based on a future noise event, a method identifies, by a first electronic device, a future noise event. The method communicates the noise mitigation alert to a second electronic device. The method further adjusts, by the second electronic device, the speaker volume concurrent with the commencement of the future noise event.

Abstract: Utterances spoken or sung by a first person can be received, in real time. The detected utterances can be compared to at least a stored sample of utterances spoken or sung by the first person. Based on the comparing, audio of the utterances spoken or sung by the first person can be isolated from a background noise. A volume of the utterances spoken or sung by a first person relative to the background noise can be determined. A key indicator that indicates the volume of the detected utterances spoken or sung by the first person relative to the background noise can be generated. Based on the key indicator, information indicating the volume of the detected utterances spoken or sung by the first person relative to the background noise can be communicated.

Abstract: A wearable audio output device in a physical environment includes an input device and one or more microphones. While ambient sound from the physical environment is being detected by the microphone(s), the wearable audio output device: while in a first mode, provides a first audio output including one or more pass-through audio components selected so as to increase audio pass-through of the ambient sound; detects an input via the input device; in response to detecting the input, and in accordance with a determination that the input is a first type of gesture, transitions from the first mode to a second mode; and, while in the second mode, provides a second audio output including one or more cancellation audio components selected so as to increase audio cancellation of the ambient sound.

Abstract: In some examples, loudness enhancement based on multiband range compression may include determining, based on variations in compression parameters that include compression thresholds and compression ratios, corresponding variations in loudness levels for a specified loudness standard. A learning model may be trained based on the variations in the compression parameters and the corresponding variations in the loudness levels. A specified loudness level for a device may be ascertained, for example, from a user of the device. The compression parameters for the specified loudness level may be determined based on the trained learning model. Further, sub-band compression of an input audio signal may be performed, based on the determined compression parameters, by processing the input audio signal using a perfect reconstruction filterbank.

Abstract: There is provided a smart audio system including multiple audio devices and a central server. The central server confirms a model of every audio device and a position thereof in an operation area in a scan mode. The central server confirms a user position or a user state to accordingly control output power of a speaker of each of the multiple audio devices in an operation mode.

Abstract: An audio playback device detects via an acoustic sensor of an audio playback device, ambient noise surrounding the audio playback device. The audio playback device updates an equalization filter based on the detected ambient noise, wherein the equalization filter adjusts one or more acoustic parameters of content presented by the audio playback device. The audio playback device adjusts a leakage control parameter based on the detected ambient noise, wherein the leakage control parameter adjusts an amount of leakage of content presented by the audio playback device. The audio playback device applies the equalization filter and the adjusted leakage control parameter to audio content. The audio playback device provides, via one or more acoustic speakers, the audio content to a user.