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.

Abstract: Embodiments of the invention relate generally to audio device and wearable computing devices to detect and characterize an ambient sound, to adjust an output volume of an audio device. More specifically, disclosed are systems, components and methods to generate audio signals associated with an audio device and an output volume, receive ambient sound signals associated with an ambient sound source, detect the ambient sound signals reaching a threshold intensity, analyzing digital data representing the ambient sound and adjust the audio signals to change the output volume according to a category of the ambient sound signals.

Abstract: An audio stream is received from one or more of a plurality of microphones. A relationship between the audio stream and a user is detected. The detection is based on the audio stream from the plurality of microphones. The user wears a wearable device that includes the plurality of microphones. A direction of the audio stream is determined. The determination is based on the audio stream from the plurality of microphones. An alert to the user wearing the wearable device is output. The output is based on the direction of the audio stream. The output is also based on the relationship between the audio stream and the user.

Abstract: A method performed by a first hearing device, includes: determining a first gain value, a second gain value, or both the first and second gain values; generating an intermediate signal including or based on a combination of the first directional input signal and the second directional input signal, wherein the first and second directional input signals in the combination are combined based on the first gain value, the second gain value, or both of the first and second gain values; and generating an output signal for the output unit based on the intermediate signal; wherein one or both of the first gain value and the second gain value are determined in accordance with an objective of making a proportion of the first directional input signal and a proportion of the second directional signal at least substantially equal.

Abstract: An active noise cancellation (ANC) circuit includes: an audio generating circuit; an audio input unit for transmitting an audio to be broadcast or a sound testing signal as a first time domain signal; an audio receiver for receiving a background sound as a second time domain signal, wherein the background sound corresponds to the audio to be broadcasted or the sound testing signal; and a channel estimation unit, configured to receive the first time domain signal and the second time domain signal, perform time-frequency conversion upon the first time domain signal and the second time domain signal to obtain a first frequency domain signal and a second frequency domain signal respectively, and generate a frequency response according to the first frequency domain signal and the second frequency domain signal. The frequency response is for adjusting an active noise cancelling coefficient of the ANC circuit to improve a noise cancellation effect.

Abstract: Disclosed is a signal processing apparatus including a surrounding sound signal acquisition unit, a NC (Noise Canceling) signal generation part, a cooped-up feeling elimination signal generation part, and an addition part. The surrounding sound signal acquisition unit is configured to collect a surrounding sound to generate a surrounding sound signal. The NC signal generation part is configured to generate a noise canceling signal from the surrounding sound signal. The cooped-up feeling elimination signal generation part is configured to generate a cooped-up feeling elimination signal from the surrounding sound signal. The addition part is configured to add together the generated noise canceling signal and the cooped-up feeling elimination signal at a prescribed ratio.

Abstract: A method comprising determining feature values of an input audio window and determining model parameters for the input audio window based on processing of feature values using a neural network.

Abstract: Methods and apparatus, including software, for collecting and managing public music performance royalties and royalty payouts are described. On the listeners side, song/audio fingerprint data is collected and transmitted to the rights owner side, where the rights owner side verifies the song/audio fingerprint data, calculates royalty payments, and in some cases, automates the royalty payments. Public music performance royalty payments are based on the song/audio fingerprint data collected by listeners/clients, as well as business logic servers.

Abstract: A computer device for configuring an alarm system comprises at least one sound measuring device and a processor configured to receive from the sound measuring device an ambient noise level at a plurality of measuring locations in a building, determine for each of the plurality of the measuring locations, a target alert sound level of an alert generated by at least one of a plurality of alerting devices based at least in part on the ambient noise level at the respective measuring location, receive an actual sound output value of the alert at each of the plurality of measuring locations from one or more of the plurality of alerting devices, and adjust a sound output of each of the plurality of alerting devices by an adjustment value based on the actual sound output values and the target alert sound levels.

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: Disclosed is a signal processor for headphone off-ear detection. The signal processor includes an audio output to transmit an audio signal toward a headphone speaker in a headphone cup. The signal processor also includes a feedback (FB) microphone input to receive a FB signal from a FB microphone in the headphone cup. The signal processor also includes an off-ear detection (OED) signal processor to determine an audio frequency response of the FB signal over an OED frame as a received frequency response. The OED processor also determines an audio frequency response of the audio signal times an off-ear transfer function between the headphone speaker and the FB microphone as an ideal off-ear response. A difference metric si generated comparing the received frequency response to the ideal off-ear frequency response. The difference metric is employed to detect when the headphone cup is disengaged from an ear.

Abstract: Embodiments of the present invention provide an earphone volume adjustment method and apparatus. The method includes: when it is detected that an intensity of external environmental noise is greater than a preset threshold, obtaining position information and/or a motion status of a user; determining a time window according to the position information and/or the motion status; and adjusting earphone volume according to an intensity of external environmental noise in the time window. According to the embodiments of the present invention, an inappropriate phenomenon such as turning up or turning down the earphone volume in a short time can be avoided, and the earphone volume can be appropriately adjusted by comprehensively considering the intensity of the external environmental noise and the position information and/or the motion status of the user. Therefore, user experience is improved.

Abstract: A computer implemented tool and method for assisting users of earphones with selecting an earbud that will provide the best fit for the user. The tool collects image data of the user's ear, along with fit data associated with the user's experience (for example, comfort and/or stability data). The tool further includes a database of ear data and associated objective/subjective data that is utilized to calculate a fit value representative of the quality of fit based on the image data and the fit criteria data generated by the user. The tool will output at least one of an indication of fit level of the earphone in the user's ear based on the fit value, and a recommendation to the user for altering the selected earphone to improve fit of the earphone within the user's ear based on the fit value.

Abstract: At least one embodiment is directed to a method for automatically activating ambient sound pass-through in an earphone in response to a detected keyword in the ambient sound field of the earphone user, the steps of the method comprising at least receiving at least one ambient sound microphone (ASM) signal; receiving at least one audio content (AC) signal; and comparing the ASM signal to a keyword and if the ASM signal matches a keyword then an AC gain is created.

Abstract: Embodiments provide techniques for controlling an audio output for a headphone system. In one embodiment, a headphone system comprising a speaker, a communication interface configured to receive a first audio signal from a first audio source hosted on a first computing device and a master audio signal from a master audio source hosted on a second computing device separate from the first computing device, wherein the first audio source and master audio source are separate from the headphone system, and an audio control module, is configured to output the first audio signal on the speaker using a first volume level, receive the master audio signal, and in response, decrease a volume of the first audio signal relative to the first volume level, output the master audio signal on the speaker at a second volume level greater than the decreased volume of the first audio signal, and after determining the master audio signal is finished, increase the volume of the first audio signal.

Abstract: A volume of sound adjustment method includes the following steps: outputting an audio with a played volume; detecting a volume of environmental noise; comparing the volume of environmental noise with a setting volume, and setting the lower one as a first target volume and setting the higher one as a second target volume; and gradually adjusting the played volume from the first target volume to the second target volume at an adjusting speed. In addition, an electronic device for adjusting a volume and a non-transitory computer readable storage medium device are also disclosed.

Abstract: A D/A converter coverts a digital audio signal into an analog audio signal. An analog volume circuit receives an output of the D/A converter. A controller controls the analog volume circuit. The controller shortens transition time at each step in the analog volume circuit further as gain is lower.

Abstract: A microphone assembly includes an acoustic transducer and an audio signal electrical circuit configured to receive an output signal from the acoustic transducer. The output signal includes an audio signal component and a tracking signal component. The audio signal component is representative of an acoustic signal detected by the acoustic transducer and the tracking signal component is based on an input tracking signal applied to the acoustic transducer. The audio signal electrical circuit includes an analog to digital converter configured to convert the output signal into a digital signal, an extraction circuit configured to separate the tracking signal component and the audio signal component from the digital signal, an envelope estimation circuit configured to estimate a tracking signal envelope from the tracking signal component, and a signal correction circuit configured to reduce distortion in the audio signal component using the tracking signal envelope.

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: A seat haptic system includes a seat comprising a plurality of actuators configured to generate a haptic output, and a controller communicatively coupled to the plurality of actuators. The controller is configured to receive a first input signal associated with first audio content at a first volume level, and control the plurality of actuators to generate a first haptic output based on the first audio content, the first haptic output having a first magnitude associated with the first volume level of the first audio content. The controller is further configured to receive a second input signal associated with second audio content at a second volume level different from the first volume level, and control the plurality of actuators to generate a transitional haptic output based on a difference between the first volume level and the second volume level.

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 system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

Abstract: In embodiments of the present invention improved capabilities are described for a computer-based method for reproducing an acoustic environment including accessing a computer stored multi-dimensional sound profile of a first space, wherein the multi-dimensional sound profile of the first space includes at least one multi-dimensional sound signature, the at least one multi-dimensional sound signature comprising a time-based sound reflection sequence within the first space for at least two sound parameters; and modifying a characteristic of an array of speakers in a second space to change the multi-dimensional sound profile in the second space to resemble at least one characteristic of the computer stored multi-dimensional sound profile of the first space.

Abstract: Systems and methods for noise cancelation in a listening area include functionality that activates a microphone of the remote control device of a home entertainment system or a microphone of another mobile device to collect a baseline white noise profile of a user's media content listening area, such as the user's family room, living room or other TV viewing space. This baseline white noise profile may be used by the user's set-top box (STB) or other receiving device to generate the corresponding noise canceling audio signal to create a noise-canceling effect and a more immersive and enjoyable listening experience for the user. A plurality of audio data samples from which to select, each representing a different baseline profile of ambient white noise associated with the media content listening area, may be generated and used for different devices that generate white noise.

Abstract: This document describes a method that includes receiving a driver signal for an acoustic transducer of an acoustic device. The method also includes receiving a signal from a microphone of the acoustic device, and, processing the driver signal through a filter to provide a reference signal. The filter has a transfer function associated with a condition of the acoustic device. The method also includes comparing the signal to the reference signal to determine whether the signal has a threshold similarity to the reference signal, and, indicating the condition of the acoustic device in respond to the determined threshold similarity.

Abstract: Disclosed is a method of controlling a personalized audio frequency equalizer by an audio device for outputting a sound source The method includes: a user's age grasping step of grasping a user's age by inquiring the user's age; a user's hearing ability measurement step of measuring, for each frequency, the minimum audible volume; a personalized equalizer creation step of creating the personalized equalizer using a system test user hearing data (HTUHD) that the user may hear when a volume having a system test frequency is outputted, and an application test user hearing data (STUHD), which is a value of volume that the user may hear, while increasing a volume having an application test frequency; and a sound source output step of outputting a sound source through the personalized equalizer created at the personalized equalizer creation step.

Abstract: Headphones include a first microphone configured to receive ambient sound at an outside of an external auditory canal of a user; a second microphone configured to receive sound inside the external auditory canal; a speaker configured to output sound toward the external auditory canal; a determiner configured to determine whether or not wind noise has occurred by comparing a first signal based on the sound received the first microphone with a second signal based on the sound received by the second microphone; and a processor configured to output, to the speaker, a signal obtained by adding an input signal to the first signal when the determiner determines that wind noise has not occurred.

Abstract: Exemplary multipath digital microphones described herein can comprise exemplary embodiments of automatic gain control and multipath digital audio signal digital signal processing chains, which allow low power and die size to be achieved as described herein, while still providing a high DR digital microphone systems. Further non-limiting embodiments can facilitate switching between multipath digital audio signal digital signal processing chains while minimizing audible artifacts associated with either the change in the gain automatic gain control amplifiers switching between multipath digital audio signal digital signal processing chains.

Abstract: The present disclosure relates to methods and devices for a hearing system. A method of retrieval of hearing device data from a hearing device of a hearing system is provided, wherein retrieval of hearing device data comprises obtaining challenge data in a server device of the hearing system; transmitting the challenge data from the server device to a user application of a user accessory device; transmitting a challenge request comprising the challenge data from the user application to the hearing device; receiving a challenge response comprising response data from the hearing device; forwarding the response data from the user application to the server device; verifying the response data in the server device based on the challenge data; and optionally deriving the hearing device data from the response data, if verifying the response data is successful.

Abstract: A volume adjustment method and a terminal relate to the field of terminal technologies and applied to adaptively adjust a volume during a voice conversation, and improve user experience. The terminal collects sound data during a voice conversation, where the sound data includes uplink voice data and background noise, and adjusts, when receiving downlink voice data, a volume of the downlink voice data based on the sound data.

Abstract: An electronic apparatus includes a microphone; a communicator configured to communicate with an external device; and a controller configured to generate sound data based on a sound input to the microphone, receive audio data from the external device through the communicator, the audio data corresponding to a sound output from the external device, and control to transmit, through the communicator, a command to the external device to adjust an intensity of the sound output from the external device based on a result of comparison between the sound data and the audio data.

Abstract: Embodiments of the present invention pertain to reducing or eliminating speech recognition error when background noise is detected at a caller's location. For example, when background noise is detected at the caller's location, the caller may be prompted to use dual-tone multi-frequency (DTMF).

Abstract: Disclosed are tone compensation device and method for an earset. A tone compensation device (method) for an earset, according to the present invention, is characterized by: extracting, in a parameter extraction unit, phase and amplitude parameters for each frequency from a voice signal transmitted from an in-ear microphone and an out-ear microphone; comparing, in a parameter comparison unit, the extracted phase and amplitude parameters for each frequency of the in-ear microphone and the extracted phase and amplitude parameters for each frequency of the out-ear microphone, respectively; and compensating, in a parameter compensation unit, for parameters having differentials between measurement values by means of the phase and amplitude parameter value for each frequency of the out-ear microphone.

Abstract: Systems and methods provide a solution to a common nuisance of loud background noise encountered when watching television, allowing the issue to be solved with little or no user interaction. Systems and methods automatically modulate device volume based on background noise level. The systems and methods record a background level of noise, and if that background level of noise increases, the systems and methods automatically increase the content volume which had been set by the user. Exceptions are made for brief sounds or for viewer conversations.

Abstract: A document editor user interface interacts with a user in receiving content from the user and provides suggestions determined via cognitive computing to the user while the user is authoring a document. A search engine searches for information associated with the content and returns the information. A matching candidate template that matches a style of the document being authored is searched for and may be inferred based on the document being authored. The suggestions may be provided based on definitions of the candidate template. A cognitive component may continuously detect behavior of the user while authoring the document and store information associated with the detected behavior. Additional suggestions may be provided based on the detected behavior.

Abstract: A head-mounted wearable device (HMWD) may incorporate bone conduction speakers (BCS) to generate audio output that is perceptible to a wearer. Due to the mechanical connection between the BCS and the rest of the device, when the output amplitude of the BCSs is too great, the air-conducted audio “leaks” and may be heard by other people. This leakage may result in eavesdropping by, or generate a nuisance to, those other people. The noise level in the environment around the HMWD is determined and used to dynamically adjust the amplitude of the audio output generated by the BCS. This adjustment improves the comfort and intelligibility of the audio output to the wearer while minimizing audio leakage to the surrounding environment.

Abstract: A solar speaker self-adjusting control system includes a photovoltaic panel, a voltage conversion device, a speaker device, an embedded processing device, and a calling device. The photovoltaic panel is for receiving solar energy and outputting a voltage signal; the voltage conversion device is connected to the photovoltaic panel for converting the voltage signal outputted by the photovoltaic panel into voltage values required by the system; the embedded processing device is connected to the calling device and speaker device for converting a voice signal transmitted from the calling device into a control current that drives the operation of the speaker device; and the calling device is embedded into a dashboard at a front end of the motor vehicle and connected to the embedded processing device installed in the dashboard at the front end of the motor vehicle through a cable. This system achieves the self-adjusting control effect of a solar speaker.

Abstract: A system and method includes a loudspeaker having a first, low-frequency driver and a second, high-frequency driver. An error microphone is disposed near the loudspeaker and receives sound output by both drivers as well as noise. An estimation of the secondary path between the drivers and the microphones is determined, and playback audio is applied to the estimation. The output of the estimation is subtracted from the output of the microphone to determine anti-noise. This anti-noise is used to modify audio data sent to the first, low-frequency driver; the audio data is sent directly to the second, high-frequency driver.

Abstract: An information processing system includes an output controller that causes an output portion to output a start condition for speech recognition processing to be performed by a speech recognition portion on sound information input from a sound collecting portion, in which the output controller dynamically changes the start condition for the speech recognition processing to be output from the output portion.

Abstract: A system includes a first earpiece having an earpiece housing configured to isolate an ambient environment from a tympanic membrane by physically blocking ambient sound, a microphone disposed within the housing and configured to receive a first ambient audio signal from the ambient environment, a processor operatively connected to the microphone wherein the processor is configured to receive the first ambient audio signal from the microphone and determine if the first ambient signal exceeds a threshold sound level, and a speaker operatively connected to the processor. In a first mode of operation the processor determines the first ambient audio signal exceeds the threshold sound level and processes the first ambient audio signal to modify the first ambient audio signal. In a second mode of operation the processor determines the first ambient audio signal does not exceed the threshold sound level and reproduces the first ambient audio signal at the speaker.

Abstract: Techniques for providing a notification to a user in a home network are provided. An example method according to the disclosure includes receiving an alert for the user, determining a location of the user, determining one or more proximate devices based on the location of the user, determining a location of one or more proximate non-interested users based on the proximate devices, determining an activity of the user, determining a preferred notification device based at least in part on the location of the user, the location of one or more proximate non-interested users, and the activity of the user, generating a notification message based at least in part on the preferred notification device, and sending the notification message to the preferred notification device.

Abstract: A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of medical device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.