Abstract: A portion of data is obfuscated by performing a bitwise XOR function between bits of the data portion and bits of a mask. The mask is generated based on the memory address of the data portion. A bitfield representing the memory address of the data portion is split into subset bitfields. Each subset then forms the input of a corresponding primary randomizing unit. Each primary randomizing unit is arranged to generate an output bitfield that appears to be randomly correlated with the input, but which may be determined from the input if certain secret information is known. The output of the primary randomizing units is input into a series of secondary randomizing units. Each secondary randomizing unit is arranged to input at least one bit of the output of every primary randomizing unit. The output of the secondary randomizing units are then combined by concatenation to form a data mask.

Abstract: The invention provides a transmitter using a chaotic signal which turns on/off a supply voltage of a chaotic signal generator in accordance with a transmitted digital data signal, without requiring a separate modulator for combining the chaotic signal and the digital data signal. The transmitter uses a chaotic signal for modulating a predetermined digital data to transmit. A chaotic signal generator turns on to generate the chaotic signal when a supply voltage is supplied and turns off when the supply voltage is cut off. A supply voltage switch supplies/cuts off the supply voltage to/from the chaotic signal generator in accordance with the digital data. Further, the supply voltage of the chaotic signal generator is supplied/cut off in accordance with the digital data so that an output from the chaotic signal generator is a modulated signal of the digital data.

Abstract: An adjustable chaotic signal generator using pulse modulation for UWB communications, and a chaotic signal generating method thereof are provided. The chaotic signal generator for UWB communications includes a plurality of pulse generators which generates pulses of different frequencies; at least one combiner which combines the pulses generated at the pulse generators; and a plurality of local oscillators which receives signals from the combiner, respectively, and generates a chaotic signal by increasing the received signals to different frequency bands. Accordingly, a plurality of users can conduct the radio communications in a specific wireless communication range at the same time by generating the chaotic signal that can be split to the multiple channels. Also, the chaotic signal generator is structured using devices integratable on an integrated circuit.

Abstract: A scrambling initial value is shared without deteriorating transmission efficiency. On the transmission side, a scrambling initial value is created based on a part of a physical layer header not scrambled, a transmission signal sequence scrambled is created by calculating an exclusive-OR operation between a scrambled sequence generated from the scrambling initial value and a transmission data sequence, and is transmitted. On the reception side, the same descrambling initial value as the scrambling initial value is created based on a part of a physical header of a reception frame, and a reception data sequence is descrambled by calculating an exclusive-OR operation between a descrambled sequence generated from this descrambling initial value and a reception signal sequence scrambled.

Abstract: Embodiments of methods and apparatus for providing a key management system for wireless communication networks are generally described herein. Other embodiments may be described and claimed.

Abstract: Provided are a key establishment method and system using commutative linear functions. In the method, a server defines a set of linear functions that use elements of a first finite field as coefficients and satisfy a commutative rule, selects a first linear function from the set, and selects a predetermined element from a second finite field. Next, the server selects a second linear function corresponding to each of nodes from the set, generates a predetermined combination function based on the first and second linear functions, generates a value of the second linear function using the selected element as a factor, and transmits the combination function and the value of the second linear function to a corresponding node. Each node receives the value of the second linear function from a server, exchanges the received values with the other nodes, computes a value using the exchanged value as a factor of the combination function, and establishes the computed value as a shared key between the nodes.

Abstract: An embedded program on an embedded device determines whether a security key has been assigned to the embedded device. If the security key has not been assigned, the embedded program uses a random number that is provided by a manufacturer of the embedded device and that is stored in memory of the embedded device to obtain the security key for the embedded device. The security key is stored in the memory of the embedded device. The security key is used to establish secure connections with other devices.

Abstract: The inventive data processing apparatus enables own memory device to store a plurality of key distribution approval data files each containing such a header data comprising a number of “link-count” data units each designating actual number of applicable contents data per decodable contents key based on an enabling key block (EKB) distribution key enciphering key (KEK) enciphered by a corresponding enabling key block (EKB) provided for by a hierarchy key tree structure. When storing a plurality of the enabling key blocks (EKB) in a memory device, such a key enciphering key (KEK) contained in an enabling key block (EKB) having a number of link-count data units is previously decoded and stored in the memory device. By way of applying the stored (KEK) when utilizing contents data, the enabling key block (EKB) processing step is deleted, whereby promoting higher efficiency in the utilization of contents data.

Abstract: A method and apparatus to reproduce sound using a mono speaker, and a stereo sound system employing the same. The sound reproducing method can include: adding a first channel signal and a second channel signal of an input stereo signal or an input mono signal; sound field filtering by adding a sound field effect to the added signals; convolving the signal of the first channel with a first HRTF in a first convolution process; convolving the signal of the second channel with a second HRTF in a second convolution process; and adding the signal output from the sound field filtering and the signals output from the first and second convolution processes to produce an output signal.

Abstract: A synthesizer for generating a decorrelation signal using an input signal is operative on a plurality of subband signals, wherein a subband signal includes a sequence of at least two subband samples, the sequence of the subband samples representing a bandwidth of the subband signal, which is smaller than a bandwidth of the input signal. The synthesizer includes a filter stage for filtering each subband signal using a reverberation filter to obtain a plurality of reverberated subband signals, wherein a plurality of reverberated subband signals together represent the decorrelation signal. This decorrelation signal is used for reconstructing a signal based on a parametrically encoded stereo signal consisting of a mono signal and a coherence measure.

Abstract: A method and system for audio decoding. A set of channel remapping equations and channel remapping coefficients are first provided. Audio samples are received, which comprise a specific number of input channels In#. The audio samples are then processed according to the channel remapping equations and channel remapping coefficients, thereby converting the input channels In# to a specific number of output channels Out#.

Abstract: An system, method, and method of encoding/decoding a multi-channel audio signal, including a decoding level generation unit producing decoding-level information that helps a bitstream including a number of audio channel signals and space information to be decoded into a number of audio channel signals, wherein the space information includes information about magnitude differences and/or similarities between channels, and an audio decoder decoding the bitstream according to the decoding-level information. Accordingly, even a single input bitstream can be decoded into a suitable number of channels depending on the type of a speaker configuration used. Scalable channel decoding can be achieved by partially decoding an input bitstream. In the scalable channel decoding, a decoder may set decoding levels and outputs audio channel signals according to the decoding levels, thereby reducing decoding complexity.

Abstract: A method and system are provided for improving the preferred listening environment of a sound system. Initially, a calibration pulse is generated from one or more rendering devices. Next, the calibration pulse is captured at a microphone attached to the rendering devices. Thereafter, one or more time delay, gain, and frequency response characteristics of the sound system are calculated using the captured calibration pulse. Based on these calculations, the time delay, gain, and frequency response characteristics of the rendering devices are adjusted respectively to cause the sound generated from the rendering devices to reach a listener's acoustic preference.

Abstract: A device and a method for integrating 3D sound effect processing and active noise control are proposed. A digital signal processor incorporates an artificial reverberator and a 3D spatial audio processor into an audio module. The audio signal is presented via an earphone. Next, a microphone embedded in the vicinity of the loudspeaker inside the headset is used to sense an external noise while playing, and feed it back to an active noise controller, which generates an anti-noise to eliminate the external noise. Therefore, the signal to noise ratio can be increased and the 3D sound field effect can be significantly enhanced. In addition, a head-related transfer function is more efficiently implemented on the basis of an interaural transfer function in the spatial audio processing to reduce the filter order lower and hence the computation loading.

Abstract: A personalized audio system for an amusement ride includes an audio selection system, a guest identification system and a ride vehicle enabled to reproduce audio selections made using the selection system in a specific seat of the ride vehicle corresponding to the seat occupied by the guest who selected the audio during that guest's ride on the amusement ride.

Abstract: Methods and apparatuses for recreating audio signals to sound as though the signals had been recorded in a different acoustic environment are provided. The methods and apparatus may include one or more inputs that receive an audio signal and an input that receives a selected acoustic environment signal, as well as processing circuitry that produces one or more output signals representative of the audio signal being played in the selected acoustic environment. The input, output and characterization signals may be processed and recorded to storage media, either individually or together. In some embodiments, the present invention interfaces with other technology and circuitry and in other embodiments the present invention is a complete stand-alone system, such as a piece of consumer electronics.

Abstract: The invention relates to an energy saving headset 100. The headset 100 comprises a power management unit 150 that is operable to reduce the power consumption of the headset 100 when a user 110 is not present. The power management unit 150 uses capacitive sensing to detect the presence of the user 110. Capacitive sensing is advantageous since it provides a flexible and reliable sensor that can accurately detect the presence or absence of a user 110 either by detecting user proximity or user contact. Moreover, in various embodiments, the sensitivity of a capacitive sensor may be adjusted to account for user movement or changes in environmental conditions, such as, for example, the presence of water, or sweat, on the headset 100 to further improve sensing reliability. The invention further relates to headsets using user presence signals based on capacitive sensing to control other functions of the headset or to control external devices to which the headset is connected, either wirelessly or by wires.

Abstract: Status changes in a hearing apparatus and in particular a hearing device should be able to be transmitted to an external data processing facility in a more rapid manner. For this purpose, provision is made for the hearing apparatus to comprise a sensor facility to capture a status of a component of the hearing apparatus. Furthermore, a status signal facility is integrated into the hearing apparatus, said status signal facility being connected to the sensor facility and being able to be used to automatically transmit a signal to the external data processing facility, when a change in the status occurs. The status of the hearing apparatus need thus not be continually queried, but a corresponding signal is actively sent outwards in the event of a status change. The battery service life of a hearing device can hereby be increased for instance.

Abstract: This invention discloses a POP noise processing circuit structure applying to an output apparatus. Therein, the output apparatus is a panel display apparatus. The POP noise processing circuit structure comprises at least one first processing unit, a second processing unit, an operational amplifier processing unit, and an audio POP noise processing unit. The first processing unit provides an audio signal to the operational amplifier processing unit, and the second processing unit provides a system signal to the audio POP noise processing unit that provides a mute signal to the operational amplifier processing unit. When the output apparatus is in a shutdown mode or a mute mode, the audio signal includes a high voltage pulse, and utilizes the circuit characteristic of the audio POP noise processing unit to eliminate the POP noise caused from the high voltage pulse.

Abstract: The potential of a smoothing capacitor for a speaker is smoothly changed by controlling the potential with use of a PWM signal and switching between outputting the PWM signal and an actual audio signal, thereby suppressing a pop sound. The scale of circuitry is reduced and cost is lowered by implementing a PWM circuit in a one-chip microcontroller. Noise is eliminated by outputting a pulse that has been divided in the PWM signal generation stage.

Abstract: An exemplary system for creating a monophonic spectrum sweeping wave file includes a parameter receiving module (12), a determining module (14) and a creating module (16). The parameter receiving module is configured for receiving parameters on a monophonic spectrum sweeping wave file to be created. The parameters basically includes a frequency distribution of the monophonic spectrum sweeping wave file as well as other parameters specifying other aspects of the monophonic spectrum sweeping wave file. The determining module is configured for determining the frequency distribution. The creating module is configured for computing a plurality of separated frequencies according to the frequency distribution, generating corresponding wave files having the plurality of separated frequencies respectively, and creating the monophonic spectrum sweeping wave file by linking the wave files linearly in order. A related method is also provided.

Abstract: A first-order crossover network having low-pass and high-pass filters to respectively drive first and second loudspeakers in a loudspeaker system is designed such that the phase difference at a crossover frequency between output signals of the first and second loudspeakers is no greater than 60 degrees, so that the output signals are at least partially in phase. Preferably, the phase difference should be about 40 degrees to create a near in-phase effect. The polarity in which the first loudspeaker is coupled to the first-order crossover network is an inverse of the polarity in which the second loudspeaker is coupled to the crossover network. Optionally, the input signals can be equalized to flatten the magnitude responses of the crossover network.

Abstract: A sound source receives an input audio signal and a sense signal depending on an external condition. The sound source has dynamic range controller that automatically adjusts the dynamic range of the input audio signal to provide the output audio signal, the adjusting depending on the sense signal. Then the sound source converts the output audio signal to ambient sound waves.

Abstract: A method and system are provided for equalizing the loudness of an audio source. Initially, the perceptual loudness level of an audio signal is measured from one or more audio sources. Next, the loudness level of the audio signal is adjusted using the perceptual loudness level. Thereafter, the audio signal corresponding to the music selections is reproduced such that the perceived loudness to a listener is the same entirely throughout a music track corresponding to the music selections.

Abstract: A disposable-type hearing aid uses a relatively large single diaphragm or a large single diaphragm subdivided into a plurality of smaller active diaphragm areas obtained using a grate-like back support plate with ridges which contact and divide the diaphragm into the several smaller active diaphragm areas. The diaphragm and a backplate are enclosed in a metal housing and are disposed proximal and parallel to a shell-like hearing aid enclosure having sound inlets. The metal housing is closed at an end opposite the sound inlets by a printed circuit board (PCB) forming an acoustical seal for a back volume of the microphone. The PCB also carries substantially all the electronic components for the hearing aid thereon. The PCB has a ground plane in contact with the housing whereby the PCB also acts as an EMI shield. An electrical connection is formed in various ways between the back support plate and the PCB during assembly of the metal housing and components with the PCB.

Abstract: An ultra directional speaker system and a signal processing method thereof are disclosed. In accordance with the present invention, the pre-distortion compensation may be applied to the input signal in real time and a signal to be modulated is subjected to a VSB modulation to minimize the distortion according to a level of the signal, and a signal difference compensation according to an envelop detection of a current signal and a signal in previous stage.

Abstract: Provided is an electrostatic transducer which is driven by a boosted driving signal by boosting a modulated signal obtained by modulating a carrier wave with an acoustic signal in an audio frequency band, the transducer including: an output transformer T which connects the electrostatic transducer to a secondary side winding thereof in parallel and boosts the modulated signal; and a resistor R and a coupling capacitance C1 connected in series to a primary side winding of the output transformer T, wherein a circuit constant of a primary side circuit of the output transformer T including a serial circuit of the resistor R and the coupling capacitance C1 and a circuit constant of a secondary side circuit of the output transformer including a self-inductance L2 and a load capacitance CL of the secondary side winding of the output transformer T are set such that a resonance frequency f0 of a circuit formed by the self-inductance L2 of the secondary side winding of the output transformer T and the load capacitance

Abstract: An electrostatic transducer includes: a class-D power amplifier that amplifies an input signal; and a low pass filter that has a plurality of pairs of inductors and capacitors, is connected to an output side of the class-D power amplifier, and serves to eliminate switching carrier components included in an output of the class-D power amplifier. An electrostatic load capacitor of the electrostatic transducer serving as a driving load is disposed at a capacitor, which is closest to the output side of the class-D power amplifier, of circuit elements forming the low pass filter, an electrostatic coupling capacitor and an output transformer are interposed between the electrostatic load capacitor of the electrostatic transducer and an inductor closest to the output side of the class-D power amplifier of the low pass filter, and a damping resistor is connected in series to a primary coil of the output transformer.

Abstract: According to one embodiment, an audio mixing device comprising an input part in which a plurality of audio data is input, a conversion part which converts the sampling frequency of all input audio data, a control part which controls such that the sampling frequency of the input audio data is converted into the preset reference sampling frequency for each conversion part, and a mixing part which mixes all audio data output from the conversion part by converting the sampling frequency into the reference sampling frequency, is provided.

Abstract: In a capacitor microphone, a diaphragm is positioned opposite to a fixed electrode for covering inner holes of a ring-shaped support, wherein when the diaphragm is deflected to approach the fixed electrode due to electrostatic attraction upon application of a bias voltage, internal stress that occurs on the diaphragm is canceled by compressive stress that is applied to the diaphragm in advance. The diaphragm is formed using a multilayered structure including a first thin film and a second thin film whose internal stress differs from the internal stress of the first thin film, thus adjusting the total internal stress thereof. The diaphragm can be formed in such a way that a center layer having a single-layered structure is sandwiched between first and second coating layers having controlled residual tensions and resistance against hydrofluoric acid.

Abstract: Protective case for an electronic device. The device is inserted into an inner case, and the earphone-jack converter plugged into the earphone jack receptacle in the bottom of the device. Both the inner case, with electronic device, and the earphone-jack converter are inserted into an outer shell through a large opening in the back of the shell. The earphone jack is threaded through an earphone jack aperture in the outer shell and plugged into the earphone-jack converter. An outer back panel attaches to the back of the inner case, thereby forming a water-tight seal between the outer shell and the inner case. The inner case has cutouts on its front face, to provide access to the operational controls on the electronic device. The material of the outer shell is soft and flexible, and allows the user to operate the electronic device while it is encased in the protective case.

Abstract: Disclosed is an earset assembly that has a housing having a microphone port and a speaker port. A microphone is enclosed by the housing and has first and second input ports. The first input port is acoustically coupled to the microphone port to detect air pressure changes of the ear of a user. A speaker is enclosed by the housing and has an output port acoustically coupled to the speaker port to broadcast sounds to the user. The output port is acoustically coupled to the second input port of the microphone so that the microphone cancels at least a portion of feedback from the sounds broadcast by the speaker and detected at the first input port of the microphone.

Abstract: A hearing aid is provided with a switch that automatically, non-manually controls at least one of inputs, filters, or programmable parameters in the presence of a magnetic field.

Abstract: The present application relates to a hearing aid, especially a hearing aid with multiple modes of operation and efficient switching between these modes of operation. The hearing aid comprises means for performing a first mode of operation, means for performing a second mode of operation, first and second quality measuring means for providing first and second quality measures corresponding to the first and second modes of operation, respectively, and selecting means for automatically selecting one of said first and second modes of operation by applying a predetermined selection criterion to the first and second quality measures, and wherein the selecting means is programmed to select the mode of operation for which the estimated quality measure is largest when the difference between the first and second quality measures exceeds a predetermined threshold value.

Abstract: A reconfigurable processing unit for a digital hearing instrument includes an IS processor module, a plurality of processing units and a crosspoint switch matrix. The IS processor module receives a hearing instrument configuration. Each of the processing modules are configured to process audio signals received by the digital hearing instrument. The crosspoint switch matrix is coupled to the IS processor module and each of the processing modules, and includes at least one crosspoint switch that is configured to route audio signals between processing modules and to combine at least two audio signals. In addition, the IS processor module uses the hearing instrument configuration to program the configuration of the crosspoint switch and thereby control how the crosspoint switch matrix routes and combines audio signals.

Abstract: A digital hearing aid enhancing a directional performance for a patient who suffers from sole-ear hearing impairments is provided, in which the digital hearing aid cell inserted into one healthy ear is electrically connected with another digital hearing aid cell called an ear cell including a microphone worn in the other troubled ear, via an external electric wire. Two digital hearing aid cells are connected with each other via signal, power, and ground wires to thereby adjust a time delay parameter and thus make a time delay effect between the two microphones, so that a wearer who wears the digital hearing aid can better hear sound which comes from the side opposing the healthy ear, that is, from the troubled ear side.

Abstract: A hearing aid has improved immunity to RF electromagnetic interference produced from wireless communications devices. A microphone receives audio signals from the environment. Audio circuitry is connected to the microphone and amplifies the audio signals. A speaker is connected to the audio circuitry and directs the audio signals into an ear canal of the user of the hearing aid. The audio connection lines connect the microphone and audio circuitry and the speaker and audio circuitry. A filter is connected into each of the audio connection lines and operative for reducing the RF coupling from a wireless communications device.

Abstract: The hearing device comprises an insertable part, which is to be inserted into an ear canal of a user of the hearing device. The insertable part may be identical with the hearing device itself, and the insertable part comprises a casing, which casing comprises a deformable part. The hearing device comprises means for changing a cross-sectional area of the deformable part, which change in the cross-sectional area is reversible. The means for changing a cross-sectional area of the deformable part may comprise at least one deformable volume containing a fixed amount of a filler. The means can be a means for changing the cross-sectional area of the deformable part through changing the axial extension of the deformable part. The means can be a volume capacity change means for changing the cross-sectional area of the deformable part through changing the capacity of a variable volume arranged within the hearing device, the deformable part being arranged on the outside of the variable volume.

Abstract: A portable speaker of a portable multimedia device includes a digital amplifier to receive a digital audio signal from the portable multimedia device and to digitally process and output the received digital audio signal to a speaker unit, and a battery to supply electric power to at least one of the digital amplifier and the portable multimedia device.

Abstract: The present invention relates to a microphone that includes a housing and a diaphragm and backplate located with the housing. The housing has a sound port for receiving the sound. The diaphragm undergoes movement relative to the backplate, which it opposes, in response to the incoming sound. The backplate has a charged layer with a first surface that is exposed to the diaphragm and a second surface opposite the first surface. The backplate further includes a conductor for transmitting a signal from the backplate to electronics in the housing. The conductor faces the second surface of the charged layer. To minimize the charge degradation created by contact with or infiltration of foreign materials, the first surface, the second surface, or both surfaces of the charged layer includes a protective layer thereon.

Abstract: A bone conduction transducer suitable for use in a listening device, such as hearing aids, in-ear monitors, headphones, electronic hearing protection devices, and very small scale acoustic speakers, has an end mass assembly disposed within the housing. The end mass assembly is mounted to the acoustic assembly and operatively coupled to the motor assembly via a coupling assembly.

Abstract: A multi-functional earphone cum microphone is disclosed. The device comprises an earphone, a microphone, a call-transferring box and transmission cable or wire, characterized in that the earphone includes a main earphone and a secondary earphone being connected by a connection wire, and from the interior of the main earphone, a specific length of earphone signal wire is extended and the end head of the signal wire is mounted with the call-transferring box having a connection male socket for the linkage with connector of any transmission source.

Abstract: A headset with retractable cord for portable devices with automatic switching capability system, comprising an electronic device, a headset, a cord extending between the electronic device and the headset, and a retracting assembly for retracting the cord into a coil during periods of non-use.

Abstract: Disclosed is a virtual multi-channel speaker unit, comprising a casing having a transmission portion through which sounds pass; a solenoid coil arranged in the casing and applied with acoustic signals; a magnetic body around which the solenoid coil is wound to form magnetic field; a diaphragm that faces one side of the magnetic body and vibrates to transmit sound waves through the transmission portion; and a transmission conduit having an inlet hole through which sound waves and compressed air generated behind the diaphragm upon the vibration of the diaphragm are introduced, and an outlet hole for discharging the sound waves and compressed air introduced through the inlet hole, the inlet hole being connected to a portion of the backside of the casing, the outlet hole being oriented toward the front of the diaphragm. The virtual multi-channel speaker unit reduces fatigue of the eardrum, improve sound quality and transmit sound close to the original sound thereof.

Abstract: A lanyard assembly can include a primary lanyard adapted to be worn around the neck of a user and a secondary lanyard positioned proximate the primary lanyard. The secondary lanyard can include a wire for providing audio transmission, a plug carried by the secondary lanyard proximate a first end of the secondary lanyard and connected to the wire, and an audio output device carried by the secondary lanyard proximate a second end of the secondary lanyard and connected to the wire. The lanyard assembly may further include a fastening arrangement. The fastening arrangement can include first and second cooperating fastening components. The first fastening component may be carried on the primary lanyard, and the second fastening component may be carried on the secondary lanyard proximate one of the plug and the audio output device. The user can selectively engage and disengage the first fastening component with the second fastening component.

Abstract: Embodiments of a speaker assembly and cooperating suspension ring designed to allow installation of the assembly in a horizontal ceiling tile from below are disclosed. The speaker assembly includes a body or housing with a flange at one end. The body has a diameter that allows it to fit through a hole the ceiling tile and the suspension ring. The assembly is further provided with clamps attached to the flange next to the body. In order to bring the flange into contact with the ceiling tile, the assembly must be aligned so that the clamps penetrate cooperating voids in the suspension ring. The assembly is then rotated in a direction depending on the tile thickness and lowered into a position where the clamps engage with projections on the suspension ring so that the suspension ring supports the weight of the assembly.

Abstract: An audio output device that has the configuration of a slat curtain includes a speaker unit provided in each slat. Therefore if the audio output device is used hanging from a window frame, ceiling, on a wall or any other such structure, the audio output device achieves a function of outputting sound while maintaining the functions of a regular slat curtain, such as the adjustment of natural lighting permitted to pass through the slat curtain and the separation of indoor spacer. Thus, a user is able listen to audio output without conscious awareness of the existence of the audio output device.

Abstract: It is an object of the present invention to provide an improved speaker device capable of restricting a wiring position of a speaker cord using a simple structure, reducing the number of parts necessary for use in speaker cord fixation. The speaker device of the present invention has a speaker unit, a frame structure having an attachment section for attaching the speaker unit to an attachment base. The frame structure has an attachment section for attaching the speaker unit to an attachment base, a fixing section (wiring position restricting means) for restricting the wiring position of the speaker cord led out from the speaker unit, on the frame structure by virtue of a damper member. The frame structure and the wiring position restricting means are formed by an identical material such as resin or metal. Therefore, it is possible to restrict the wiring position of the speaker cord using a simpler structure.

Abstract: A loudspeaker includes a movable diaphragm, a resilient centering device for centering and guiding the movement of the diaphragm, and a magnet system for controlling the movement of the diaphragm, where the diaphragm is positioned between the magnet system and the resilient centering device.

Abstract: According to one embodiment, a speaker comprises a frame, a drive source attached to the frame, which converts an electric signal into mechanical vibration, a diaphragm coupled to the drive source, which emits an acoustic wave corresponding to the mechanical vibration, and an attachment structure coupled to a predetermined portion (field unit attachment face) of the frame, which has a flange at a position remote from the predetermined portion. The attachment structure has predetermined compliance in a direction of vibration of the diaphragm, and which has predetermined stiffness in a direction substantially orthogonal to the direction of vibration of the diaphragm.