Abstract: A diaphragm for a loudspeaker may include a continuous primary diaphragm having an upper surface and a lower surface, where the primary diaphragm comprises a ring-shaped, flat region having an inner diameter and an outer diameter. The diaphragm may also include a reinforcing ring attached to the upper surface of the primary diaphragm, where the reinforcing ring has an inside diameter and an outside diameter, and where the reinforcing ring is attached to the upper surface of the primary diaphragm such that the inside diameter of the reinforcing ring coincides with the inner diameter of the flat region.

Abstract: A speaker assembly including a frame and a magnet assembly positioned within the frame. The magnet assembly may include a magnet and a top plate. The assembly further including a sound radiating surface suspended over the magnet assembly. The sound radiating surface includes a flexible circuit. A suspension suspending the sound radiating surface over the magnet assembly is further provided. The suspension may be over molded to the sound radiating surface and the frame. A voice coil extends from a bottom side of the sound radiating surface and electrically connects to the flexible circuit.

Abstract: A driver for a loudspeaker including an inner magnet disposed within a housing, an outer magnet disposed within the housing radially outward of the inner magnet, an inner voice coil having a first diameter, an outer voice coil having a second diameter, the second diameter being larger than the first diameter, where poles of the inner and outer magnets are oppositely disposed, where the inner magnet is configured to contribute only to a magnetic circuit of the inner voice coil, and where the outer magnet is configured to contribute to the magnetic circuit of the inner voice coil and to a magnetic circuit of the outer voice coil.

Abstract: Disclosed is an acoustic transducer for converting a sound signal into an electric signal, including a mobile element movable under the effect of the sound signal, a fixed element opposite the mobile element, a recess, and a dissipative element between the mobile and fixed elements. The coupled system has a natural frequency corresponding to a resonance frequency of the transducer set at maximum sensitivity. The mobile element, the fixed element, the dissipative element and the recess are configured so the quality factor of the acoustic transducer>2. The recess has a straight prismatic, cylindrical, or frustoconical shape, the mobile element forming a first base of the prism, cylinder, or frustum, the fixed element being inside the prism, cylinder or frustum, over the second base of the prism, cylinder or frustum. Such a transducer also incorporates an analogue filtering function for filtering the signal around the natural frequency of same.

Abstract: Backplates for MEMS devices. In one embodiment, the backplate includes an interconnect layer, a first layer, a second layer and a plurality of openings. The interconnect layer includes a first side and a second side that is opposite from the first side. The first layer is coupled to the first side of the interconnect layer. The second layer is coupled to the second side of the interconnect layer. The plurality of openings are located between a first side of the backplate and a second side of the backplate.

Abstract: A fixed electrode configured to face a diaphragm and forming a capacitance with the diaphragm, including a plurality of protrusions formed by plastic deformation on one surface of the fixed electrode that is to face the diaphragm, so as to protrude toward the diaphragm.

Abstract: A method (300) of operating a hearing aid system (100), wherein the dynamic range of input signal levels is improved by reducing the sensitivity of an input transducer in response to a trigger event while at the same time applying a gain adapted to compensate the reduced sensitivity and a hearing aid system (100, 200) adapted to carry out the method.

Abstract: A method (400) of operating a hearing aid system, wherein the sensitivity of an input transducer is reduced in response to a trigger event, whereby clipping due to limited dynamic range of the input transducer can be avoided and wherein the sensitivity of the input transducer is controlled using a digital pulse train whereby a high resolution control is obtained and a hearing aid system adapted to carry out the method.

Abstract: A method (300) of operating a hearing aid system wherein the acoustical output signal intensity levels are confined to a range that primarily high-spontaneous rate auditory nerve fibres respond to, hereby providing sound processing that may benefit individuals with an auditory neurodegeneration, a computer-readable storage medium having computer-executable instructions, which when executed carries out the method, a hearing aid system (100, 200) adapted to carry out the method and a method of fitting a hearing aid system.

Abstract: The exemplary disclosure describes a hearing system e.g. comprising a Hearing Aid device comprising a cellphone and/or user worn device where some of the programs are carried out by components embedded onto the user worn device and some programs by hearing system components, e.g. which are inherently part of cellphones. The hearing system improves the intelligibility of voice messages arriving e.g. through the cellphone and/or other speaker, and/or e.g. via connected earphones and/or directly through the free air. The user can call diverse programs suitable for different situations, by using e.g. inertial sensors embedded in the hearing system, e.g. in the user worn system and/or e.g. are inherently part of the cellphone.

Abstract: A loudspeaker module for a hearing device has a loudspeaker, which has a loudspeaker diaphragm and a drive for the loudspeaker diaphragm, and a housing, in which the loudspeaker is arranged. Furthermore, the loudspeaker module contains an antenna unit, which has an antenna coil having a coil axis, a tubular coil core, which forms a sound channel, and an antenna base plate, in which a sound passage opening that opens into the sound channel is formed. The antenna coil, the coil core and the antenna base plate in this case prescribe an antenna characteristic of the antenna unit. The side wall of the housing on the diaphragm side is in this case formed by the antenna base plate.

Abstract: Method for providing sound to at least one user, involves supplying audio signals from an audio signal source to a transmission unit; compressing the audio signals to generate compressed audio data; transmitting compressed audio data from the transmission unit to at least one receiver unit; decompressing the compressed audio data to generate decompressed audio signals; and stimulating the hearing of the user(s) according to decompressed audio signals supplied from the receiver unit. During certain time periods, transmission of compressed audio data is interrupted, and instead, at least one control data block is generated by the transmission unit in such a manner that audio data transmission is replaced by control data block transmission, thereby temporarily interrupting flow of received compressed audio data, each control data block includes a marker recognized by the at least one receiver unit as a control data block and a command for control of the receiver unit.

Abstract: The present disclosure relates to an RF antenna adapted to receive and/or transmit electromagnetic RF signals within a first frequency range enclosing a first frequency of resonance of the RF antenna, the RF antenna comprising: an electrically conductive antenna element having a feed for electrically connecting to an RF transmitter and/or an RF receiver; an electronic component adapted to receive and/or provide one or more electric signals from/to an electronic circuit within a second frequency range not overlapping the first frequency range; and one or more electric leads electrically connected to lead the one or more electric signals between the electronic component and the electronic circuit, each of the one or more electric leads being electrically connected to the electronic circuit through a respective inductor adapted to reflect and/or attenuate signals within the first frequency range and pass signals within the second frequency range.

Abstract: An audio processing device has an information extractor that extracts identification information from a first audio signal in a first frequency band that includes an audio component of a sound for reproduction and an audio component including the identification information of the sound for reproduction and a signal processor that generates a second audio signal that includes the identification information extracted by the information extractor and that is in a second frequency band higher than the first frequency band, with a sound represented by the second audio signal being emitted from a sound emission device.

Abstract: A derivation unit (240) determines, as a specified rhythm component of a musical piece, a rhythm component detected within a predetermined time range including a time of reception of tap timing information TAP and derives a first frequency band the spectrum intensity of which is equal to or greater than a predetermined value. The derivation unit (240) also determines, as an unspecified rhythm component, a rhythm component detected outside the predetermined time range including the time of reception of the tap timing information TAP and derives a second frequency band the spectrum intensity of which is equal to or greater than the predetermined value. Thereafter, a calculation unit (250) calculates a third frequency band, which is included in the first frequency band and which does not include the second frequency band, and then transmits, to a filter unit (260), a passed-frequency designation BPC that designates the third frequency band.

Abstract: A portable audio device may be configured to measure load characteristics of headphones. The device may measure direct current (DC) and/or alternating current (AC) characteristics of the load. These characteristics may be measured by an audio component, such as an audio codec chip or integrated circuit (IC) controller, and reported to software or firmware executing on a processor coupled to the audio component. The software or firmware may then take action based on the measured load characteristics. For example, the load characteristics may be compared to a database of headphones and their known load characteristics to determine a particular headphone model or type of headphone attached to the audio output. The processor may then apply an appropriate equalization curve.

Abstract: An electronic device may be configured to identify a load coupled to the device. The device may measure direct current (DC) and/or alternating current (AC) characteristics of the load to identify the load. The device may then take action based on the identification of the load. For example, a specific transducer may be identified as coupled to the electronic device and an appropriate equalization curve applied to an audio output of the device. The measurement of characteristics of the load may include controlling a reference generator according to a search algorithm, such as a step ramp or binary search, to identify the load. An analog-to-digital converter (ADC) may operate through the search algorithm to provide feedback to digital circuitry regarding how to proceed through the search algorithm to identify the load.

Abstract: An electronic device may be configured to identify a load coupled to the device. The device may measure direct current (DC) and/or alternating current (AC) impedances of the load to identify the load. The device may then take action based on the identification of the load. For example, a specific transducer may be identified as coupled to the electronic device and an appropriate equalization curve applied to an audio output of the device. The measurement of load impedance may include controlling a reference generator according to a search algorithm to identify the load, including compensating the measured impedance for temperature changes. An analog-to-digital converter (ADC) may operate through the search algorithm to provide feedback to digital circuitry regarding how to proceed through the search algorithm to identify the load.

Abstract: Examples described herein involve providing playback device calibration user interfaces to guide a calibration process for one or more playback devices in a playback environment. In one example, a network device receives audio samples continuously from a microphone of the network device for a predetermined duration of time, wherein the predetermined duration of time comprises a plurality of periodic time increments. At each time increment within the predetermined duration of time, the network device dynamically updating on a graphical display of the network device, (i) a representation of a frequency response based on audio samples that have been received between a beginning of the predetermined duration of time and the respective time increment, and (ii) a representation of the respective time increment relative to the predetermined duration of time.

Abstract: A headset having corresponding computer-readable media comprises: a first portion, wherein the first portion comprises a microphone; a second portion, wherein the second portion is rotatably coupled with the first portion; one or more sensors, wherein at least one of the sensors is configured to sense a current difference angle between the first portion and the second portion; a controller configured to generate a message responsive to the current difference angle being outside a range of difference angles, wherein the range of difference angles is associated with acceptable performance of the microphone; wherein the headset is configured to provide the message responsive to the controller generating the message.

Abstract: Systems and methods disclosed herein include, while a microphone of a first networked microphone device is enabled, determining whether a first reference device is in a specific state, and in response to determining that the first reference device is in the specific state, disabling the microphone of the first networked microphone device. Some embodiments further include, while the microphone of the first networked microphone device is enabled, receiving a command to disable the microphone of the first networked microphone device via one of the microphone of the networked microphone device or a network interface of the networked microphone device, and in response to receiving the command to disable the microphone of the networked microphone device via one of the microphone of the networked microphone device or the network interface of the networked microphone device, disabling the microphone of the networked microphone device.

Abstract: Microphone systems and methods of determining absolute sensitivities of a MEMS microphone. The microphone system includes a speaker, a MEMS microphone, and a controller. The speaker is configured to generate an acoustic pressure. The MEMS microphone includes a capacitive electrode, a piezoelectric electrode, and a backplate. The capacitive electrode is configured to generate a first capacitive response based on the acoustic pressure and generate a first mechanical pressure based on a capacitive control signal. The piezoelectric electrode is coupled to the capacitive electrode. The piezoelectric electrode is configured to generate a first piezoelectric response signal based on the acoustic pressure and generate a second piezoelectric response signal based on the first mechanical pressure.

Abstract: In at least one embodiment, a system for monitoring movement of a wireless microphone that transmits an audio signal on a stage is provided. The system includes a plurality of antennas for being positioned on stage and each being positioned on a different zone of the stage and each being configured to wirelessly receive an audio signal from a wireless microphone. The system further includes a controller that is operably coupled to each antenna. The controller is configured to determine the signal strength for the audio signal received at each antenna at least two or more times and to determine a signal strength trend for each antenna in response to determining the signal strength for the audio signal at each antenna at least two or more times.

Abstract: A first signal may be received indicative of audio to be played by a speaker. A second signal may be received which comprises (i) a voice input received by a microphone and (ii) at least a portion of the audio played by the speaker at a same time that the microphone receives the voice input. Based on the first signal, nonlinearities output by the speaker which played the audio may be determined. At least the nonlinearities from the second signal may be removed to output a third signal comprising substantially the voice input received at the microphone.

Abstract: The object is to provide a display system of mobile audio devices which enables comparative display between frequency range of audio source read externally and memorize by the main body of the mobile audio devices and frequency range replayed by the main body of the mobile audio devices. The display system of audio devices comprises the audio source file 18 memorizing outer audio source data, the replay unit 14 of audio source data of the audio source file 18, the device data memory 16 of data of replay devices, the controller 15 controlling among circuits, the sampling rate output 21 of audio source data of said audio source file 18, the replay sampling rate output 23 outputting said replay sampling rates, the audio source output display part 12 displaying said sampling rate output 21, and the replay output display part 13 displaying said replay sampling rate output 23.

Abstract: A method generates a sound localization point (SLP) where binaural sound externally localizes to a person during a telephone call. The method determines coordinates of a location of a handheld portable electronic device (HPED) with respect to the person and saves this location as the SLP. During the telephone call, a voice of another person is convolved so the voice externally localizes to the person as the binaural sound at the SLP.

Abstract: Techniques of rendering audio involve applying a balanced-realization state space model to each head-related transfer function (HRTF) to reduce the order of an effective FIR or even an infinite impulse response (IIR) filter. Along these lines, each HRTF G(z) is derived from a head-related impulse response filter (HRIR) via, e.g., a z-transform. The data of the HRIR may be used to construct a first state space representation [A, B, C, D] of the HRTF via the relation .G(z)=C(zI?A)?1B+D This first state space representation is not unique and so for an FIR filter, A and B may be set to simple, binary-valued arrays, while C and D contain the HRIR data. This representation leads to a simple form of a Gramian Q whose eigenvectors provide system states that maximize the system gain as measured by a Hankel norm. Further, a factorization of Q provides a transformation into a balanced state space in which the Gramian is equal to a diagonal matrix of the eigenvalues of Q.

Abstract: A method, a computer readable medium, and an apparatus for multichannel audio playback adaptation. A user input defining a listening region relative to a loudspeaker arrangement is received via an input. A distance calculating unit determines minimal distances between the listening region and the loudspeakers. A comparator compares the determined minimal distances with a critical distance. A parameter setting unit then adapts output parameters for those loudspeakers for which the determined minimal distance is below the critical distance.

Abstract: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. A vector of coefficient domain signals is separated into a vector of coefficient domain signals having a constant number of HOA coefficients and a vector of coefficient domain signals having a variable number of HOA coefficients. The constant-number HOA coefficients vector is transformed to a corresponding spatial domain signal vector. In order to facilitate high-quality coding, without creating signal discontinuities the variable-number HOA coefficients vector of coefficient domain signals is adaptively normalised and multiplexed with the vector of spatial domain signals.

Abstract: A signal mixing architecture dealing with signal spatialization and distributing signals to more than two destinations that provides a single axis of pan control where a user can move a signal across N-number of channels. Inverse panning and complex pan choreography can be achieved from a simple set of controls in addition to standard mixer functions.

Abstract: The present technology relates to a sound processing apparatus and method, and a program for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

Abstract: A sound field control system SY according to the present invention is provided with: a sound emission unit 210 which causes a plurality of speakers to emit a test sound; a sound pickup unit 130 which picks up the test sound using a microphone; an analysis unit 150 which compares information indicating a sound emission timing of a test signal sequence for causing the speakers to successively emit the test sound at a prescribed timing with a sound pickup timing of each test sound that has been picked up, and which calculates a time difference between the sound emission timing of each test sound and the sound pickup timing; and a signal processing unit 230 which performs, based on the calculated time difference between the sound emission timing of each test sound and the sound pickup timing, a delay process for a voice signal supplied to each speaker.

Abstract: Provided is an apparatus and method for frontal audio rendering in interaction with a screen size, the method including measuring playback environment information used to play back input content; and correcting an audio signal to be output based on the measured playback environment information and production environment information included in the input content.

Abstract: A method, a computer readable storage medium, and an apparatus for determining a target sound scene at a target position from two or more source sound scenes. A positioning unit positions spatial domain representations of the two or more source sound scenes in a virtual scene. These representations are represented by virtual loudspeaker positions. A projecting unit then obtains projected virtual loudspeaker positions of a spatial domain representation of the target sound scene by projecting the virtual loudspeaker positions of the two or more source sound scenes on a circle or a sphere around the target position.

Abstract: Methods and systems are provided for ultrasonic emitter based audio devices. An example audio device may comprise one or more ultrasonic emitters. The audio device may generate audio outputs that comprise ultrasonic signals, with the ultrasonic signals emitted using the one or more ultrasonic emitters. The directionality of emission of the ultrasonic signals may be based on a position of a listener and/or a location of the at least part of the listener's body relative to the audio device. The ultrasonic signals and/or the positioning of the one or more ultrasonic emitters may be adjusted based on the position of the listener and/or the location of the at least part of the listener's body to optimize the directionality of emission of the ultrasonic signals. The position of the listener and/or the location of the at least part of the listener's body may be determined in real-time, such as using sensory information.

Abstract: The aspects disclosed herein are related to providing a stereoscopic audio environment that is based on speaker position and/or orientation. Once the speaker (or groups of speakers) are identified by the above-described techniques, the audio signal uniquely delivered to each of the speakers may be customized to produce an optimal sound environment.

Abstract: A method comprising: displaying a spatio-temporal graph that visually indicates the positions of a sound object at each time in a range of times and, in addition, simultaneously indicates a state of the sound object over the range of times; enabling user adaptation of the spatio-temporal graph to change a position of the sound object during at least a portion of the range of times and/or a state of the sound object over at least a portion of the range of times; and enabling rendering of spatial audio output using the position and state of the sound object indicated by the adapted spatio-temporal graph.

Abstract: Aspects of the subject disclosure may include, for example, identifying a configuration of a second end user device that includes device data and applications, generating a configuration template associated with the second end user device, providing a first end user device with access to the configuration template for enabling the first end user device to be configured, detecting an adjustment to the configuration of the second end user device, and transmitting a notification of a configuration change at the second end user device responsive to the detecting of the adjustment to enable the first end user device to be reconfigured according to the adjustment to the configuration of the second end user device. Other embodiments are disclosed.

Abstract: The present invention provides a method for data transmission in a machine type communication device terminal, comprising steps of: performing the following steps when the terminal receives cell common message: A. receiving first indication information from an EPDCCH on a first resource block, the first indication information indicating a second resource block for transmitting the cell common message; B. receiving the cell common message from a PDSCH on the second resource block; performing the following steps when the terminal receives downlink data: I. receiving second indication information from EPDCCH on a third resource block; IL detecting whether DCI information transmitted to the terminal exists in the second indication information, and if the DCI information transmitted to the terminal exists, II-1. decoding information of a fourth resource block in the DCI information; II-2. receiving the downlink data transmitted from the base station to the terminal from the PDSCH on the fourth resource block.

Abstract: Systems and methods for shipping management in a network of moving things. As non-limiting examples, various aspects of this disclosure provide systems and methods for integrating sensors, mobile access points, and/or the network of moving things with shipping container management systems (e.g., port management systems, train management systems, trucking management systems, etc.).

Abstract: A WSN has multiple sensor nodes, a bi-directional gateway sensor node, and a server. The WSN also includes circuitry configured to process a route request message from an origination sensor node to a destination sensor node. A message packet of each of the sensor nodes includes a minimum energy field, a minimum energy harvesting rate field, and a sum of energy field. The circuitry is also configured to determine a new delivery route based upon comparing a minimum energy and a minimum energy harvesting rate of an instant sensor node with a minimum energy and a minimum energy harvesting rate contained in the message packet, and select a lower minimum energy and a lower minimum energy harvesting rate from the instant sensor node or the message packet. The circuitry is also configured to broadcast the lower minimum energy and the lower minimum energy harvesting rate throughout the WSN.

Abstract: A system for mounting in a vehicle and for providing a control signal to a remote device based on information stored in a portable electronic device includes a radio frequency transmitter for transmitting a control signal to the remote device. The system also includes a first circuit configured to receive first information from the portable electronic device via inductive-coupling between the portable electronic device and the first circuit when the portable electronic device is brought within the induction field of the first circuit. The system also includes a second circuit configured to use the first information received from the portable electronic device and to at least one of format the control signal in accordance with the first information and to cause the radio frequency transmitter to format the control signal in accordance with the first information.

Abstract: A computing system that includes wearable computing components may be provided that creates an interactive game for at least two players. Finite state machines may be provided with a range of states that correspond to an array of possible states of a player of the game. Each player is assigned a client device, which preferably may be embedded in wearable computing components. The client device uses software to execute a first instance of a finite state machine. The client deceives are connected in a network with server devices. The state machines transition states based upon player interaction with other players and objects. When the state machines transition states, the players are notified and their respective states within the game also change.

Abstract: Methods and systems for using near field communication (NFC) protocol and logic to calibrate welding operations and systems are described. Further, methods and systems for using NFC logic and tags are described for networking, calibrating and linking components that comprise welding systems.

Abstract: A server device is provided that provides a user with a mode for assigning newer privileges. The server device includes: a communication unit for receiving, from a terminal device, position information indicating a position of the terminal device in the real world, the terminal device being held by a user, and a first application being executed in the terminal device; a memory unit for storing a plurality of items stored in correspondence with prescribed locations in the real world; and a control unit for assigning to the user a privilege usable in a second application different from the first application, on the basis of one or more items that are selected by the user from the plurality of items stored in the memory unit and that are arranged in correspondence with locations within a prescribed range from the real-world position indicated by the position information.

Abstract: A user enters a location with a user device. A beacon device broadcasts a first beacon device code comprising a hardware identifier via a local wireless network at the location. A service application of the user device receives the first beacon device hardware identifier, logs a check-in status of the user, and transmits the check-in status to a service provider system. The beacon device generates, after a predetermined period of time, a subsequent beacon device code comprising a random number to broadcast at the location via the local wireless network. In response to receiving the subsequent beacon device code broadcast by the beacon device, the user device logs and transmits a subsequent check in status to the service provider system via the network. The service provider system provides services to the user device or another device at the location in accordance with the check-in status of the user device.

Abstract: It is presented a method for determining a position of a wireless device within a single radio cell served by a plurality of remote radio heads. The method is performed in a position determination node and comprises the steps of: receiving a measured power signature comprising at least two, in time separated, power measurement values of a downlink signal, the measured power signature originating from the wireless device; comparing the measured power signature with a set of reference power signatures, wherein each one of the plurality of remote radio heads is associated with a reference power signature and the reference power signatures differ for at least two of the plurality of remote radio heads; and determining a position in that the wireless device is in the vicinity of one of the at least one remote radio head whose reference power signature best matches the measured power signature.

Abstract: Techniques for supporting location services for a home Node B (HNB) and its user equipments (UEs) are disclosed. In an aspect, location services may be supported for a UE by having an HNB inter-work between user plane and control plane location solutions. In one design, the HNB receives a request for a location service for the UE and communicates (i) with a location server via the user plane location solution and (ii) with the UE via the control plane location solution to support the location service for the UE. The HNB inter-works between the user plane and control plane location solutions. In another aspect, a location server may be used to support assisted GNSS (A-GNSS) for HNBs and UEs. In one design, an HNB exchanges PCAP messages with the location server via an HNB GW and exchanges RRC messages with a UE to support a location service for the UE.

Abstract: A method for distributed beacon management, including: broadcasting a packet from the beacon; receiving the packet at an application executing on a plurality of user devices; and transmitting beacon information extracted from the packet to the management system from the application.

Abstract: Methods and systems are disclosed relating to proximity detection of devices and targeted information services, such as information campaigns. Disclosed methods provide targeted campaigns to a user, such as targeted campaigns to presentation devices based on proximity of user devices near the information devices. Other methods provide targeted campaigns across various applications, while other methods provide for scheduling campaigns to premises device identifiers based on an event schedule, and other methods provide a hierarchal system for monitoring different sets of premises device identifiers when a user device can monitor for a limited number of premises device identifiers at a time.