Abstract: A feedforward ambient noise reduction arrangement includes, within a housing, a loudspeaker device for directing sound energy into an ear of a listener. Disposed externally of the housing, and positioned to sense ambient noise on its way to the listener's ear, are plural microphone devices capable of converting the sensed ambient noise into electrical signals for application to the loudspeaker to generate an acoustic signal opposing the ambient noise. Importantly, the overall arrangement is such that the acoustic signal is generated by said loudspeaker means in substantial time alignment with the arrival of said ambient noise at the listener's ear.

Abstract: An acoustic protective cover assembly can include a retracted membrane material traversing across an acoustic pathway defined by an opening in a housing. The retracted membrane material can include a fibril structure characterized by serpentine fibrils.

Abstract: A system and method for performing liquid volume determination in a smart environment, the method comprising involve determining that a liquid is being poured. The method includes identifying the liquid, and determining the liquid volume of the liquid being poured based on sound analysis of audio sensed by a microphone. A scenario in which the liquid is being poured is analyzed, and an action is determined based on the scenario. The action includes triggering an actuator of the smart environment, obtaining additional information about the liquid, or issuing an alert.

Abstract: An apparatus including a body, a plurality of microphones arranged in a predetermined geometry relative to the body such that the apparatus is configured to capture sound substantially from all directions around the body to produce direction and ambience information for the captured sound, and electronics for processing signals from the plurality of microphones.

Abstract: A microphone apparatus is provided. The microphone apparatus includes a microphone array and an integrated circuit. The microphone array includes at least three microphones arranged in a straight line with a non-uniform configuration. The integrated circuit is electrically connected to the microphone array. The integrated circuit is configured to process a merged sound signal from different combinations of microphones using a time-domain filter to generate an output sound signal. The sound gain of the output sound signal at each sound-receiving angle of each frequency is substantially uniform.

Abstract: Embodiments described herein generally relate to analyzing a signal generated by a device under test (DUT). In particular, the signal generated by the DUT may be compared to a reference signal to determine pass/fail results for the DUT. For example, a method may include: storing, on a computing device, a reference signal from a reference device; receiving a test signal from a device under test (DUT); synchronizing the reference signal and the test signal based on a time-synchronization buffer of each signal; after the synchronization, comparing the test signal and the reference signal to determine a pass or fail result for the DUT; and generating a notification indicating the pass or fail result for the DUT.

Abstract: The disclosure includes a headset including one or more earphones and a connector configured to couple data and charge between the headset and a user equipment (UE). The headset also includes a charge node. The charge node includes a charge port for receiving UE charge from a charge source. The charge node also includes a downstream port for coupling audio data toward the earphones. The charge node further includes an upstream port for coupling the audio data toward the earphones via the downstream port and coupling UE charge from the charge port toward the UE via the connector.

Abstract: An apparatus and a method for drawing attention to an object are provided. The apparatus includes an information acquisition unit configured to acquire personal information of a person, a processor that determines a target object based on the personal information and identifies positional information of the target object, and an emitter for emitting at least one of ultrasound waves and light to the target object based on the positional information. The method includes the steps of acquiring personal information of a person, determining the target object based on the personal information and identifying positional information of the target object, and emitting at least one of ultrasound waves and light from an emitter to the target object based on the positional information.

Abstract: A method of controlling charging of a wireless headphone device includes estimating a usage time value that corresponds to an anticipated future occurrence of usage of the wireless headphone device and estimating a resume time value that corresponds to a time for charging to commence to allow for completion of charging from a first predetermined state of charge value to a second predetermined state of charge value by the usage time value. The method also includes determining that the wireless headphone device has been placed in a charging case. A battery of the wireless headphone device is charged to the first predetermined state of charge value and then enters a deactivated mode, wherein the charging case is configured to return the wireless headphone device to an activated mode according to the resume time value.

Abstract: Audio rendering devices comprising at least one audio rendering unit for playing audio streams, a processing circuit and a wireless communication unit for establishing wireless links and related method are disclosed. One audio rendering device is configured to receive a remote audio stream prioritization policy comprising one or more lists of associations between audio stream attributes and priority values, update a local audio stream prioritization policy based on the remote audio stream prioritization policy, identify audio streams available via wireless links, retrieve audio stream attributes of the available audio streams, select an available audio stream to be played among the available audio streams, based on the updated local audio stream prioritization policy and based on the audio stream attributes of the available audio streams, and play the selected audio stream, by the audio rendering unit of the audio rendering device.

Abstract: A virtual-reality device displays a virtual scene. The device generates audio data for an area source in the virtual scene. The area source is within a predefined near-field distance from the listener. The device selects sample points from the area source and determines, for each sample, energy contributions to two respective successive shells of spherical shells that extend from the listener to the predefined near-field distance. The two shells enclose the sample point source, and the determined energy contributions correspond to sound originating from the sample point source. For each shell, the device determines a head-related impulse response (HRIR) by combining the determined energy contributions for that shell. The device determines an overall HRIR for the virtual scene by combining the determined HRIRs for the shells and combines the audio data with the overall HRIR. The device transmits the combined audio data to sound-producing elements of the device.

Abstract: A display apparatus includes: a display panel configured to be rolled or unrolled, at least one vibration module on a rear surface of the display panel, the at least one vibration module being configured to: be rolled or unrolled with the display panel, and vibrate the display panel, and a partition on the rear surface of the display panel, the partition being spaced apart from and surrounding the at least one vibration module, wherein the at least one vibration module includes: a plurality of first portions having a piezoelectric property, and a plurality of second portions between the plurality of first portions, the plurality of second portions having flexibility.

Abstract: An earpiece (100) and acoustic management module (300) for in-ear canal echo suppression control suitable is provided. The earpiece can include an Ambient Sound Microphone (111) to capture ambient sound, an Ear Canal Receiver (125) to deliver audio content to an ear canal, an Ear Canal Microphone (123) configured to capture internal sound, and a processor (121) to generate a voice activity level (622) and suppress an echo of spoken voice in the electronic internal signal, and mix an electronic ambient signal with an electronic internal signal in a ratio dependent on the voice activity level and a background noise level to produce a mixed signal (323) that is delivered to the ear canal (131).

Abstract: In one embodiment, an audio system can generate parametrically formulated noise signals. According to an embodiment, an audio system can both determine the hearing ability of an individual and increase the hearing ability of the individual by using parametrically formulated noise. According to an embodiment, an audio system can generate parametrically formulated noise having a power spectrum amplitude that is a function of an individual's hearing threshold across a range of frequencies as measured using parametrically formulated noise test signals.

Abstract: A method for voice recording and an electronic device thereof are provided. The method includes determining a voice recording mode from among a plurality of voice recording modes, determining a voice beamforming direction according to the determined voice recording mode, and recording voice signals based on the determined voice beamforming direction.

Abstract: A system for controlling an engine tone by an artificial intelligence based on a sound quality index of a vehicle may include a sound output device for generating a reinforcing sound to reinforce an engine sound of the vehicle; an engine characteristic measurement sensor for measuring sound source characteristics of the engine sound; an interior noise measurement sensor for detecting interior noise of the vehicle; a signal processing controller that receives signals from the engine characteristic measurement sensor in real time and controls the sound output device such that the engine sound reaches a target tone; and a tone control operation unit connected to the signal processing controller to optimize the sound quality index such that the engine sound reaches the target tone through the artificial intelligence.

Abstract: The present invention is a communication system for use in debates, for example, in Congress, Parliament, Presidential elections, online meetings, webinars, etc. The system includes multiple single-channel, time-controlled microphones, each having a STOP button. The STOP button activates the next speaker's microphone and timer while turning off the timer and microphone of the current speaker. The microphones can only be activated by the host or by the STOP button of another microphone. This prevents speakers from interrupting one another. If the time limit is reached, STOP button-activated-control is relinquished to the next speaker and his/her timer and microphone is activated. The remaining and/or elapsed time of each speaker may be displayed on computer screens of each participant, on a display for the audience and/or on lighted digital displays and/or dials. Time limits and the sequence of speeches may be altered by the host.

Abstract: Systems and methods for manual characterization of perceived transducer distortion are described. The system includes a signal generator, a controller, a multi-band compressor, and an audio transducer. The signal generator is configured to generate a test signal for a frequency band. The test signal includes at least two simultaneous frequency-modulated tones, which may be combined together in an amplitude-modulated envelope. The at least two simultaneous frequency-modulated tones have different frequencies within the frequency band. The controller is configured to receive user input indicating a gain value. The multi-band compressor is coupled to the signal generator. The multi-band compressor is configured to adjust an amplitude of a component of the test signal based on the gain value. The audio transducer is coupled to the multi-band compressor. The audio transducer is configured to generate an audio signal based on the test signal.

Abstract: According to an embodiment, a microspeaker includes an acoustic micropump structure configured to pump at a first frequency above an upper audible frequency limit. The acoustic micropump structure is further configured to generate an acoustic signal having a second frequency by adjusting the pumping. Adjusting the pumping includes adjusting a direction of pumping for the acoustic micropump structure according to the second frequency. Adjusting the direction of pumping includes changing a direction of flow of an elastic medium through the acoustic micropump structure from a first direction to a second direction. The second frequency is below the upper audible frequency limit.

Abstract: A terminal assembly structure of a MEMS microphone, including a signal let out board disposed at a terminal and a silicon microphone disposed at the signal let out board. The silicon microphone includes a housing, a substrate forming an accommodation space with the housing, and an MEMS chip accommodated in the accommodation space. A position of the substrate corresponding to the MEMS chip is disposed with a sound inlet connected to the outside, wherein a position where the signal let out board corresponding to the silicon microphone is disposed with an accommodation hole. The housing is accommodated in the accommodation hole. The substrate abuts a surface of the signal let out board and covers the accommodation hole. A surface of the substrate disposed with the housing is provided with a pad which is electrically connected with the signal let out board.