Abstract: Wireless communication schemes and techniques are described, wherein a secondary device is configured to eavesdrop information communicated between a source and a primary device. Secondary device transmits a NACK signal to jam ACK signals from the primary device to the audio source, forcing a retransmit of audio information from the source to the primary, and available over an eavesdropping link between the secondary device and the source.

Abstract: Method for outputting an audio signal (S) into an interior (4) forming part of a passenger compartment (2) of a motor vehicle (3) via an acoustic output device (7) comprising a left and a right acoustic output channel (5, 6), comprising the steps: supplying an audio signal (S), splitting of the supplied audio signal (S) into a plurality of acoustic audio signal components (S.1-S.3) dependent on the direction of perception, mixing of each acoustic audio signal component (S.1-S.3) dependent on the direction of perception to a right and/or a left output channel (5, 6) of an acoustic output device (7) comprising a right and a left acoustic output channel (5, 6), output of the acoustic audio signal components (S.1-S.3) dependent on the direction of perception via the right and/or the left acoustic output channel (5, 6) of the acoustic output device (7).

Abstract: A personalized HRTF is generated from a generic HRTF by having a person don headphones, then select a generic HRTF model from a list. Test signals emulating sounds from a source at a specific virtual speaker location are played on the headphones, and the person indicates where he thinks the emulated sounds are coming from. Any difference between the virtual speaker location and the “heard” location is used to adjust the parameters (amplitude, phase, and frequency response) of the generic HRTF to render a HRTF personalized to the listening person.

Abstract: According to various embodiments of the disclosure, an electronic device includes a first speaker, a second speaker, at least one sensor that detects a state of at least one of the first speaker and the second speaker and at least one processor electrically connected to the first speaker, the second speaker and the at least one sensor, wherein the at least one processor outputs an audio signal through the first speaker and the second speaker, identifies states of the first speaker and the second speaker by using the at least one sensor, and decreases a volume level of an audio signal output by the first speaker and increases a volume level of an audio signal output by the second speaker when it is determined that the state of the first speaker is an abnormal state. In addition, various embodiments understood from the specification are possible.

Abstract: A diaphragm for use is a loudspeaker is described. This diaphragm may include a housing with an elongated shape having a length along a first axis that is longer than a width along a second axis. Moreover, the housing may include: an outer surface and an inner surface; an outer opening defined by an outer edge and an inner opening defined by an inner edge; and regions having heights relative to the inner surface, where the regions are grouped in pairs that are positioned equidistant and symmetrically about the inner opening along the first axis. Furthermore, the regions may have a second length along the first axis, and the second length may be less than a distance along the first axis between the outer edge and the inner edge. Note that the regions may increase a stiffness of the diaphragm relative to a stiffness of a material in the diaphragm.

Abstract: Determination of a set of acoustic parameters for a headset is presented herein. The set of acoustic parameters can be determined based on a virtual model of physical locations stored at a mapping server. The virtual model describes a plurality of spaces and acoustic properties of those spaces, wherein the location in the virtual model corresponds to a physical location of the headset. A location in the virtual model for the headset is determined based on information describing at least a portion of the local area received from the headset. The set of acoustic parameters associated with the physical location of the headset is determined based in part on the determined location in the virtual model and any acoustic parameters associated with the determined location. The headset presents audio content using the set of acoustic parameters received from the mapping server.

Abstract: Techniques for user-adaptive volume selection are disclosed. A system receives audio input that includes a user request to a virtual assistant. The system determines one or more sound characteristics corresponding to the audio input, and applies the sound characteristic(s) to a volume selection model. The volume selection model selects a volume level based on the sound characteristic(s). The system plays an audio response to the user's request at the selected volume level. Subsequently, the system receives user input to modify output volume to a different volume level. Based on the user input, the system calibrates the volume selection model. When the system receives another audio input, the system analyzes the other audio input to determine one or more sound characteristics of the other audio input. The system uses the calibrated volume selection model to select a volume level for an audio response to the other audio input.

Abstract: An acoustic signal processing device 1 includes: a focal point position determination unit 12 that obtains a plurality of sets of initial focal point coordinates, coordinates of the virtual sound source, and a direction of directivity thereof, and for a pair of sets of initial focal point coordinates with different polarities among the plurality of sets of initial focal point coordinates, multiplies the sets of initial focal point coordinates by a rotation matrix based on the coordinates of the virtual sound source to thereby determine sets of focal point coordinates, the rotation matrix being specified from the direction of the directivity; a circular harmonic coefficient conversion unit 13 that calculates weights to be applied to multipoles including the sets of focal point coordinates from a circular harmonic coefficient; a filter coefficient computation unit 14 that, for each of the speakers in the speaker array, computes a weighted driving function to be applied to the speaker from the sets of focal poin

Abstract: Provided are a display apparatus and a computing apparatus including the same. A display apparatus includes: a display module including a display panel configured to display an image, a vibration plate on the display module, a first air gap between the vibration plate and the rear surface of the display module, a vibration module on the vibration plate, and a system rear cover on the vibration plate, a second air gap between the system rear cover and the vibration plate, wherein the vibration plate includes a communication part configured to allow air to flow between the first air gap and the second air gap.

Abstract: A system configured to perform directional speech separation using three or more microphones. The system may dynamically associate direction-of-arrivals with one or more audio sources in order to generate output audio data that separates each of the audio sources. Using three or more microphones, the system may separate audio sources covering 360 degrees surrounding the microphone array, whereas a two-microphone implementation is limited to 180 degrees. The system identifies a target direction for each audio source, dynamically determines directions that are correlated with the target direction, and generates output signals for each audio source. The system may associate individual frequency bands with specific directions based on a phase difference detected by two or more microphones.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for dynamic volume adjustment via audio classification. Examples methods include analyzing, with a neural network trained model, a parameter of an audio signal associated with a first volume level to determine a classification group associated with the audio signal, determining an input volume of the audio signal, the selection based on the classification group associated with the audio signal, applying a gain value to the audio signal, the gain value based on the classification group and the input volume, the gain value to modify the first volume level to a second volume level, and applying a compression value to the audio signal, the compression value to modify the second volume level to a third volume level that satisfies a target volume threshold.

Abstract: Methods and systems for verification of an environmental seal provided by an encapsulant coating of a bottom-ported MEMS microphone package. A purposeful acoustic leak is provided on an upper surface of a package housing (in the form of an additional acoustic port) and a sealing material is applied to an outer surface of the package housing. A properly applied encapsulant coating will completely seal the additional acoustic port on the upper surface of the package housing. However, the placement of the additional acoustic port on the upper surface of the package housing will have a significant, detectable effect on the frequency response of the microphone if it is not completely sealed by the encapsulant coating. Accordingly, the environmental seal provided by the encapsulant coating is verified by confirming, based on the acoustic frequency response testing, that the encapsulant coating has effectively sealed the additional acoustic port on the upper surface of the package housing.

Abstract: Examples associated with interference generation are described. One example system includes a personal digital assistant. A listening device in the system may receive voice input to control the personal digital assistant. An interference generator integrated with the listening device may generate a noise pattern to prevent the listening device from receiving voice input. An interface for the interference generator may allow a user to activate and deactivate the interference generator and provide a visual indicator describing whether the interference generator is active.

Abstract: Embodiments of wireless audio systems and methods for wirelessly communicating audio information are disclosed herein. In one example, a wireless audio system includes a first wireless headphone and a second wireless headphone. The first wireless headphone is configured to receive, from an audio source, audio information using a first type of short-range wireless communication; and in response to successfully receiving the audio information from the audio source, transmit an audio information message including the audio information using a second type of short-range wireless communication different from the first type of short-range wireless communication.

Abstract: The present image forming apparatus, in response to reception of facsimile data, outputs a monitoring sound at a predetermined sound volume, and displays a setting input screen for adjusting the sound volume on a display unit.

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: In general, the present invention relates to a method and system for synthesizing artificial reverberation using modal analysis of a room or resonating object. In one embodiment of the inventive system, a collection of resonant filters is employed, each driven by the source signal, and their outputs summed. With filter resonance frequencies and dampings tuned to the modal frequencies and decay times of the acoustic space or resonating object being simulated, and filter gains set according to the source and listener positions within the space or object, any number of acoustic spaces and resonant objects may be simulated.

Abstract: A system and method for performing perceptually-transparent RTF estimation for a two-channel system with low estimation errors. The system includes two loudspeakers for the left and right channels positioned in a room and a single microphone, positioned in the vicinity of a listener in the room. The system performs spectral splitting using a filter bank and a complementary filter bank, to obtain separation in frequency between the two channels. Calibration signals being uncorrelated across the two channels are, generated using the original audio signals and the constructed filters. Then, the calibration signals are played back by the loudspeakers, instead of playback of the original audio signals. At the end, the system records the playback of the calibration signals by the microphone and simultaneously estimates the two-channel RTF using the recorded played back calibration signals.

Abstract: Two subsystems, each including a microphone, a speaker, a canceling sound-generating adder, an error-computing adder, and two adaptive filters and two auxiliary filters that accept two noises as input, are provided in correspondence with two cancellation positions. Each canceling sound-generating adder adds together the outputs from the adaptive filters and outputs the result to the speaker of each subsystem. Each error-computing adder adds together the output from the microphone of the subsystem and the output from the auxiliary filter of the subsystem, and the result is treated as the error of the adaptive filters of each subsystem. A transfer function is learned in advance and set in each auxiliary filter such that each error computed by each error-computing adder becomes zero (0) when a transfer function in which each noise is canceled at each cancellation position in a predetermined standard acoustic environment is set in each adaptive filter.

Abstract: Provided are a directional acoustic sensor that detects a direction of sound, a method of detecting a direction of sound, and an electronic device including the directional acoustic sensor. The directional acoustic sensor includes a sound inlet through which a sound is received, a sound outlet through which the sound received through the sound inlet is output, and a plurality of vibration bodies arranged between the sound inlet and the sound outlet, in which one or more of the plurality of vibration bodies selectively react to the sound received by the sound inlet according to a direction of the received sound.