56results about How to "Flat Frequency Response" patented technology

A thin film bulk acoustic wave (BAW) resonator structure and filter which can be fabricated by inexpensive manufacturing techniques and in smaller size than conventional such products are to be provided. The BAW resonator structure and filter have a substrate, a first BAW resonator placed over the substrate, an acoustic reflection layer placed over the first BAW resonator and a second BAW resonator placed over the acoustic reflection layer, and the acoustic reflection layer is electroconductive. Herein, the acoustic reflection layer constitutes a first electrode, and this first electrode electrically connects and acoustically separates the first BAW resonator and the second BAW resonator.

The invention discloses a channel balance and sound field control method and device of a digitalized speaker system on the basis of a single generalized encoder. The method comprises the following steps of: (1), digital format conversion; (2) multi-channel balance processing; (3) sound field control processing; (4) serial-parallel conversion processing; (5) generalized encoding processing; (6) extraction selecting processing; (7) serial-parallel conversion processing; and (8) power amplification of digital signals of all channels, and driving of sound production of a digitalized speaker load. The device comprises a sound source, a digital format converter, a channel balancer, a sound field controller, a serial-parallel converter, a generalized encoder, an extraction selector, a serial-parallel converter, a multi-channel digital power amplifier and the digitalized speaker load; and all the units are connected in sequence. According to the method and device disclosed by the invention, volume, power consumption, cost and hardware implementation resources of the system are reduced, the electroacoustic conversion efficiency and the anti-jamming capability of the system are enhanced, and the inband frequency response plainness degree and the sound field control capability of the system are increased.

The present invention relates to a suspension member for a diaphragm and to an electro-acoustic transducer applying such suspension member. The suspension member comprises an inner portion being adapted to be attached to a piston, an outer portion being adapted to be coupled to a substantially stationary portion of an associated electro-acoustic transducer, and a flexible member connecting the inner and outer portions. The flexible member comprises at least one electrically conducting element adapted to transport electrical signals across the suspension member.

An apparatus includes a multi-layer printed circuit board having a first through-hole via for a signal connection and a second through hole via for power / ground connections. The printed circuit includes a transmission line connected to at least one through-hole via. A resistor is connected between the first and second through-hole vias to eliminate a resonance notch and achieve a flat frequency response for insertion loss.

A voice coil assembly capable of realizing a flat thin loudspeaker having a high efficiency, reduced divided vibrations, a flat frequency response, and reduced operation defects. The voice coil assembly includes a plurality of internal-winding voice coils, each including a rectangular bobbin having a rectangular cross section and defining a rectangular space therein and an internal rectangular coil fixed to an inner wall surface of the rectangular bobbin defining the rectangular space, wherein an outer wall surface of the rectangular bobbin of one internal-winding voice coil is adhered and fixed to an outer wall surface of the rectangular bobbin of another internal-winding voice coil.

A test probe tip constructed substantially from resistive material. The resistive material is made of resistive conducting material substantially enclosed in and dispersed throughout encapsulating material. The test probe tip has a probing end for probing electronic circuitry and a connection end for interfacing with a probing head. The resistive conducting material forms at least one path through the encapsulating material from the probing end to the connection end. The resistive conducting material may be a plurality of longitudinally-extending resistive / conductive members or a plurality of particulate resistive / conductive members.

A method of converting an audio signal into vibratory modulation of a fluid includes converting a series of pulses representative of the audio signal into a plurality of signals having an intermediate peak-to-peak voltage; summing said signals having said intermediate voltage to provide a driver signal having a high peak-to-peak voltage; supplying said driver signal to an electrostatic fluid accelerator; and generating a corona discharge inducing said vibratory modulation of said fluid.

The invention relates an electrostatic speaker system comprising—a high voltage switching power amplifier,—an extraction filter having an input coupled to an output of the high voltage switching amplifier, and—an electrostatic speaker element having a capacitive load and an input coupled to an output of the extraction filter. The combination of the extraction filter and capacitive load form a filter circuitry having at least a first filter stage and a second filter stage. The first filter stage comprising a RLC circuit having a resonant frequency ωO and a quality factor Q>½ and wherein the second filter stage being a low pass filter comprises at least one electrical element for damping a signal component at the resonant frequency of the RLC circuit at the output of the extraction filter.

In a stalk-type field probe, each of three resistive dipoles is formed with a resistive feedline on a common circuit board. Each circuit board comprises an elongated feedline section that extends through a cylindrical stalk, and a cross-arm that is disposed at a 54.7° angle relative to the elongation of the feedline section. The feedline sections are arranged to form an equilateral triangular prism, and the cross arms are arranged so that the dipole on each cross-arm is perpendicular to an imaginary plane to which the lines along which the elements of both of the other dipoles extend are parallel.

The planar speaker contains a frame, a diaphragm, a driver, a surround, and a number of flexible suspension elements. The diaphragm is suspended in a front opening of the frame by the surround. The driver is positioned at the center of the frame and a voice coil of the driver is attached to a back surface of the diaphragm. Two beams are arranged in parallel between two opposing longer wall segments of the frame on either side of the driver. The suspension elements are supported by the two beams, respectively, and attached to a back surface of the diaphragm. The symmetric deformation of the diaphragm along its longer side is thereby constrained by the suspension elements and the deformations at the two end zones of the diaphragm vibrating at a phase opposite to that of the center of the diaphragm would be suppressed.

An apparatus and method allow receivers to quickly acquire a pseudorandom noise signal. A receiver advantageously detects frequency shifts using a compact parallel process hardware implementation of a Discrete Fourier Transform (DFT). The simultaneous detection of multiple frequencies allows the receiver to search the frequency range of the transmitted signal in larger increments of frequency, thereby increasing the speed of acquisition. One receiver does not use coherent integration before computation of the transform and advantageously maintains a flat frequency response. The flat frequency response of the DFT circuit enables searching of multiple frequency offsets without CPU intensive processing to compensate for frequency response variations. A receiver can include a Doppler correction circuit, which permits correlation data with frequency shift in the code to be non-coherently integrated among relatively fewer addresses or tap positions in memory.

An apparatus and method allow receivers to quickly acquire a pseudorandom noise signal. A receiver advantageously detects frequency shifts using a compact parallel process hardware implementation of a Discrete Fourier Transform (DFT). The simultaneous detection of multiple frequencies allows the receiver to search the frequency range of the transmitted signal in larger increments of frequency, thereby increasing the speed of acquisition. One receiver does not use coherent integration before computation of the transform and advantageously maintains a flat frequency response. The flat frequency response of the DFT circuit enables searching of multiple frequency offsets without CPU intensive processing to compensate for frequency response variations. A receiver can include a Doppler correction circuit, which permits correlation data with frequency shift in the code to be non-coherently integrated among relatively fewer addresses or tap positions in memory.

The invention discloses an improvement method for the low-speed performance of a permanent magnet synchronous motor, belonging to the field of high-performance servo control. The improvement method comprises the following steps: carrying out filtering pre-processing on the output of a speed calculation module by virtue of an improved Butterworth low-pass filter; establishing a composite PMSM mathematical model, and designing a general speed equation by the composite PMSM mathematical model; establishing a standard speed equation by the general speed equation; designing a disturbance observer with regard to the standard speed equation; adjusting the gain matrix L of the disturbance observer, correcting a lumped disturbance estimated value and compensating a speed PI adjustor through forward feedback to obtain final output. High-frequency noise components in the output of the calculation module are effectively filtered by the improved Butterworth low-pass filter, thus avoiding influence on the low-speed performance of the motor; the lumped disturbance is estimated through the disturbance observer and is subjected to forward feedback processing, so that not only can the influence of external disturbance be eliminated, but also the load torque item, the friction torque item and the inertia perturbation of the permanent magnet synchronous motor can be effectively processed, and the low-speed performance of the permanent magnet synchronous motor can be improved.

The invention discloses a method for compensating the amplitude of an emission signal of a broadband sonar system. The method comprises the following steps that a sonar emission system emits an ideal chirp signal x0[n], and an ideal sonar reception system is used to collect a distorted chirp signal y[n] in a soundproofing water pool; an error function e[n] is constructed, an estimated value, which satisfies the condition of min{E[e2[n]]}, of a transfer function of a broadband matching network and an energy transducer is searched, and a transfer function w[n] of an amplitude compensator is obtained; the amplitude compensator adjusts the amplitude of the ideal chirp signal; the power of the adjusted chirp signal is amplified; the power-amplified chirp signal is output to the energy transducer via the broadband matching network; the energy transducer converts an electrical signal into an acoustic signal and radiates the acoustic signal to water; and the ideal sonar reception system is used to collect the acoustic signal emitted by the energy transducer in the soundproofing water pool. The method can be used to reduce amplitude distortion of the emission signal, and has the advantage of high precision.

An integrated digital vehicle sound equipment system is disclosed. Vehicle sound equipment is arranged behind a headrest assembly of a rear-row seat, and the front side of the sound equipment is at a 0-90-degree included angle to the horizontal plane. The vehicle sound equipment is provided with a bar-shaped housing with a semi-elliptic cross section. The housing is divided into a left sound box and a right sound box, which are symmetric, through a baffle plate, and a digital audio control system is arranged in the housing. The integrated digital vehicle sound equipment system has the advantages of small size, small occupied area, lossless installation, dismounting convenience, easy operation, flexible application and high fidelity sound quality.

The invention relates to a silicon-based MEMS array loudspeaker. The silicon-based MEMS array loudspeaker comprises a substrate and multiple MEMS micro-acoustic units; the multiple MEMS micro-acoustic units are arrayed on the substrate; each MEMS micro-acoustic unit comprises a vibrating diaphragm, an upper insulation sheet, a lower insulation sheet, an upper conductive plate and a lower conductive plate; the vibrating diaphragm is used as an intermediate electrode; the upper insulation sheet and the lower insulation sheet are connected to the upper and lower surfaces of the vibrating diaphragm; the upper conductive plate is used as an upper electrode; the lower conductive plate is used as a lower electrode; the upper conductive plate is installed on the upper insulation sheet; the lower conductive plate is installed on the lower insulation sheet; the upper insulation sheet and the lower insulation sheet are each provided with a centre hole; the upper conductive plate is provided with multiple pronunciation holes communicated with the centre hole; and the lower conductive plate is provided with a ventilation hole communicated with the centre hole. The silicon-based MEMS array loudspeaker disclosed by the invention has the characteristics of being thin in thickness, good in sound quality and the like.

A transmitting system based on visible light communication is characterized by comprising a common collector amplifying circuit, an analog pre-equalizer, an LED current driving circuit and a light emitting diode, wherein the common collector amplifying circuit is connected with the analog pre-equalizer, the analog pre-equalizer is connected with the LED current driving circuit, and the LED currentdriving circuit is connected with the light emitting diode. A third resistor and a first capacitor is connected between the input end and the output end of the pre-equalizer after connected in parallel, and a fourth resistor is connected between the output end of the pre-equalizer and the ground. The passive analog pre-equalizer is inserted between active circuits formed by the common collector amplifying circuit and the LED current driving circuit. The design of a common base structure is used behind the analog pre-equalizer, the input impedance is very small, the frequency response of the analog pre-equalizer is ensured not to tend to be flat at a high frequency band too early, the frequency response of the LED can be effectively compensated and corrected, and the modulation bandwidth and the data transmission rate of the system are improved.

The invention discloses a pickup method based on a microphone array, a pickup device and a storage medium. The pickup method based on a microphone array comprises the following steps that A, determining the microphone array, providing the microphone array with a plurality of microphone channels, wherein each microphone channel comprises at least one microphone unit; B, establishing an error probability model as follows: representing an actual steering vector; C, constructing an optimal problem of a differential beam design algorithm based on the error probability model; D, solving an optimal solution of the optimal problem to obtain weight vectors of a plurality of microphone channels of the microphone array; and E, adjusting the voice signal collected by the microphone array according to the weight vector, and outputting the optimized voice signal. The pickup method provided by the invention has good anti-interference performance, especially has good directivity at a low-frequency position, and is low in distortion degree.

An audio signal processing apparatus comprises a receiver (403) receiving an audio signal sampled at a first sampling frequency, the audio signal having a maximum frequency below half the first sampling frequency by a first frequency margin. A filter bank (405) generates subband signals for the digital audio signal using overlapping sub-filters. A first frequency shifter (407) applies a frequency shift to at least one subband of the set of subbands and a decimator (409) decimates the subband signals by a decimation factor resulting in a decimated sampling frequency being at least twice a bandwidth of each of the overlapping sub-filters. The frequency shift for a subband is arranged to shift the subband to a frequency interval being a multiple of a frequency interval from zero to half the decimated sample frequency. The subband may be individually processed and the processed subbands may subsequently be combined to generate a full band output signal.

The invention discloses a distortionless sound playback system for counter electromotive force signals. The distortionless sound playback system is characterized by comprising a counter electromotive force signal loudspeaker and a counter electromotive force signal amplifier, wherein when the loudspeaker body is distorted, and the motion speed of a paper disk is deviated from a desired value under the pressure of air in a sound box or the influence of sound reflection, resonance, standing waves and harmonic factors, a sound coil in the counter electromotive force signal loudspeaker, which is connected with the paper disk, can instantly detect the generated deviation and feed back the deviation to the counter electromotive force signal amplifier; and the counter electromotive force signal amplifier instantly makes a response to rectify the deviation until the deviation completely disappears. The distortionless sound playback system is rational in design; the loudspeaker is changed into a standard four-end network provided with an input and an output; and moreover, an output signal can really reflect an electric signal of the motion of the loudspeaker, so that a power amplifier and the loudspeaker belong to a real electronic network system. The frequency response is wide and flat, and the sound is not subjected to distortion and phase shift; the counter electromotive force signal loudspeaker can replace all treble, alto and bass loudspeakers in the sound box; and therefore, the step of adjusting the troublesome frequency divider is omitted, and the manufacturing cost of the sound box is greatly reduced.

The invention relates to the technical field of microphone sound head, in particular to a sound head polar plate of microphone, which includes a polar plate as well as a step through hole, a blind hole, a screwed hole, a chute, a screwed blind hole and an insulating post arranged on the polar plate; the central part of the polar plate is provided with a central hole; the surrounding of the central hole is provided with a plurality of through holes which are uniformly distributed; the polar plate is characterized in that the appearance of the polar plate is elliptical; the length of the long axis of the polar plate is 45mm-11mm and the length of the short axis of the polar plate is 44mm-10mm; the diameter of the through hole is 0.7-1.3mm; the through holes are arranged with mutually symmetrical central horizontal line and vertical line of the polar plate; compared with the prior art, the sound head polar plate of elliptical microphone has simple and novel structure, wide frequency response, flat frequency response, higher frequency response characteristics and directivity, and changes the sound head structure of the previous traditional microphone. Besides, the sound head polar plate of elliptical microphone is simple to be manufactured with obvious effects, and is worthy of popularization and application.