54results about How to "Reduce reverberation" patented technology

A piezoelectric micromachined ultrasonic transducer comprising a substrate and a first dielectric film formed on the substrate. An opening having a sidewall is formed through the substrate and first dielectric film. A bottom electrode is formed on the first dielectric film spanning the opening. A piezoelectric element is formed on the bottom electrode. A second dielectric film surrounds the piezoelectric element. A conformal insulating film is formed on the sidewall of the opening. A conformal conductive film is formed in contact with the bottom electrode and on the sidewall of the opening, wherein an open cavity is maintained in the opening. A top electrode is formed in contact with the piezoelectric element.

A piezoelectric micromachined ultrasonic transducer comprising a substrate and a first dielectric film formed on the substrate. An opening having a sidewall is formed through the substrate and first dielectric film. A bottom electrode is formed on the first dielectric film spanning the opening. A piezoelectric element is formed on the bottom electrode. A second dielectric film surrounds the piezoelectric element. A conformal insulating film is formed on the sidewall of the opening. A conformal conductive film is formed in contact with the bottom electrode and on the sidewall of the opening, wherein an open cavity is maintained in the opening. A top electrode is formed in contact with the piezoelectric element.

The present invention is directed to a telephone equipped with multiple microphones that provides improved performance during operation of the telephone in a speaker-phone mode. For example, the multiple microphones can be used to improve voice activity detection, which in turn, can improve echo cancellation. In addition, the multiple microphones can be configured as an adaptive microphone array and used to reduce the effects of (i) room reverberation, when a near-end user is speaking, and / or (ii) acoustic echo, when a far-end user is speaking.

A system and method facilitating signal enhancement utilizing mixture models is provided. The invention includes a signal enhancement adaptive system having a speech model, a noise model and a plurality of adaptive filter parameters. The signal enhancement adaptive system employs probabilistic modeling to perform signal enhancement of a plurality of windowed frequency transformed input signals received, for example, for an array of microphones. The signal enhancement adaptive system incorporates information about the statistical structure of speech signals. The signal enhancement adaptive system can be embedded in an overall enhancement system which also includes components of signal windowing and frequency transformation.

The invention relates to a whole image sound barrier for an urban quick lane and a tunnel portal. The sound barrier is characterized by consisting of gate-type arc steel arch racks arranged in the upper space of a pavement, and a sound barrier main body framework consisting of rigid tie rods for connecting the adjacent arch racks and cross supporting rods, wherein purlines are arranged on the main body framework; and meanwhile, the purlines arranged on the main body framework are used as supports, anti-collision wall contact faces are formed symmetrically on the left and right sides along the central axis of the sound barrier till the pavement, and a sound absorber, a ventilated exhaust part, a sound absorber, a lighting part, a sound absorber and a lighting part are arranged at intervals, so that a full-closed for semi-closed whole image sound barrier structure capable of totally isolating direct sound is formed for a road. The whole image sound barrier for the urban main road can effectively reduce the traffic noise of the urban main road, effectively reduce the reverberation in a sound barrier area and ensure the safety and the comfort degree of a road user, and is a sound barrier suitable for a building dense area nearby the urban main road.

The invention discloses an extinction method of voice reverberation based on Wiener filtering, belonging to the technical field of voice signal processing. The method of the invention includes the four steps of input processing, reverberation analysis, reverberation processing and output processing, wherein, the input processing is to carry out sampling, endpoint detection and framing to a voice signal, the reverberation analysis is to carry out a first deconvolution operation to an onset voice and a reverberation signal thereof, so as to obtain a room impact response message of reverberation environment; the reverberation processing is to carry out filtering to the reverberation signal based on the room impact response message, namely, a second deconvolution operation, so as to obtain a dereverberation signal; the output processing is to carry out voice reconstruction, digital-to-analog conversion and signal transmission to the dereverberation signal, as well as record a dereverberation evaluation index. The method of the invention has good dereverberation effect, lower computation intensity and better timeliness.

The invention relates to an acoustic positioning system, in particular to an acoustic positioning system of a shallow sea wave detector for secondary positioning of a seabed wave detector in shallow sea petroleum earthquake exploitation. The system mainly comprises a carrier-borne part and a seabed part, wherein the carrier-borne part consists of processing software, a host machine and a receiving and transmitting combined transducer; and the seabed part consists of a plurality of seabed transponders. Because the technology for manufacturing the receiving and transmitting combined transducer is adopted, the designed source level and the sensitivity meet the positioning requirement of a complex exploitation area of the shallow sea; and the underwater working posture of the transducer is kept and the turbulence and reverberation are reduced by designing streamline turning vanes. Therefore, the system meets the requirements of high recovery ratio and high positioning efficiency of secondary positioning of a secondary positioning earthquake exploitation water detector in the Bohai Bay area.

A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients and modifies the adaptive coefficients based on a feedback output. A feedback component provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. Optionally, the system can further include a linear prediction (LP) analyzer and / or a LP synthesis filter. The system can enhance signal(s), for example, to improve the quality of speech that is acquired by a microphone by reducing reverberation. The system utilizes, at least in part, the principle that certain characteristics of reverberated speech are measurably different from corresponding characteristics of clean speech. The system can employ a filter technology (e.g., reverberation reducing) based on a non-linear function, for example, the kurtosis metric.

The present invention is a desktop speakerphone having a base-station and a detachable microphone pod. The base-station includes standard telephone components, as well as a wireless receiver and a housing for a detachable microphone pod. The detachable pod contains at least one microphone and a wireless transmitter. When the pod is attached to the base-station, and the conference mode of operation is activated, the pod microphone's audio signal goes directly to base-station audio circuitry via a wired connection. When the pod is detached and the conference mode activated, the pod microphone's audio signal now goes via the pod's wireless transmitter to the base-station's wireless receiver. This detached, wireless mode allows the microphone to be positioned anywhere in the room, thereby improving the quality of transmitted speech by increasing the speech-signal-to-room-noise ratio, and lessening the potential for room echo by reducing the acoustic coupling between base-station loudspeaker and pod microphone.

The present invention provides a time-reversal emission focusing reverse beam forming method based on high frequency time-reversal array. The beam forming method uses high frequency time-reversal array. A sound wave is emitted to irradiate the object and an echo signal from the object is received. The received echo signal is turned on time and emitted back. The time-reversal signal can focus at the object place, which performs as time waveform compression and space-focusing. An enhanced object echo signal is received again to form a receiving focusing beam. The present invention provides the time-reversal emission focusing reverse beam forming method based on the high frequency time-reversal array which uses a high frequency time-reversal mirror which is suitable for shallow water acoustic fields, reduces the reverberation and improves the detection ability to shallow water.

The invention relates to a magnesium base composite material, comprising magnesium base metal and nano reinforcing phase dispersed in the magnesium base metal. The weight percentage content of the nano reinforcing phase in the magnesium base metal is from 0.01% to 2%. The magnesium base composite material is used for manufacturing the shell of a sounding device. The invention also relates to a preparation method of the magnesium base composite material.

The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.

The invention provides a loudspeaker array sound reinforcement system for inhibiting howling. The system comprises an input signal processing module (1), a filter module (2), a multichannel power amplifier (3) and a loudspeaker array (4); the filter module (2) comprises a parameter setting unit (201), a filter group generation unit (202) and a filter group processing unit (203); the filter group generation unit (202) is used for calculating convolution matrixes Hm and Hn formed by time domain impulse response at each control point of the loudspeaker array (4) according to parameters set by the parameter setting unit (201); calculating matrixes Rb and Rd accordingly; calculating a matrix Q according to an average amplitude response fluctuation parameter expression of an equilibrium control point in a sound reinforcement target area; solving a vector w according to Rb, Rd and Q to ensure a maximum value of time domain acoustic energy contrast control of single point equilibrium; converting the vector w into a filter form; and acquiring filter parameters of the filter group processing unit (203).

Equipment (10) for reconstructing a recorded sound field includes a sensing arrangement (12) for measuring the sound field to obtain recorded data. A signal processing module (14) is in communication with the sensing arrangement (12) and processes the recorded data for the purposes of at least one of (a) estimating the sparsity of the recorded sound field and (b) obtaining plane-wave signals to enable the recorded sound field to be reconstructed.

To provide a loudspeaker broadcasting system and a loudspeaker broadcasting apparatus which can perform a loudspeaker broadcasting of appropriate contents at each block without reducing sound clearness. Bi-directional speakers 1 which performs sound-reinforcement having high intelligibility, and fire detecting sensors 3 which detect fire occurrence in a tunnel 10a, 10b and send fire occurrence signals are placed in the tunnel 10a, 10b in the tunnel extending direction. In an emergency such as a fire, emergency broadcasting is performed through each bi-directional speaker 1 at each of the predetermined blocks 5 in the tunnel 10a, 10b. The emergency broadcasting is different in contents at each block, and it is alternately performed at the adjacent blocks 5. Further, the frequency components having a long reverberation time is removed from the loudspeaker broadcasting through the bi-directional speaker 1.

A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients, the adaptive filter modifying at least one of the adaptive coefficients based on a feedback output. The invention further includes a feedback component that provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. The invention further provides a noise statistics component that stores noise statistics associated with a noise portion of an input signal and a signal+noise statistics component that stores signal+noise statistics associated with a signal and noise portion of the input signal. The invention further provides a spatial filter that provides an output signal based, at least in part, upon a filtered input signal, the filtering being based, at least in part, upon a weighted error calculation of the noise statistics and the signal+noise statistics.

The invention relates to an earphone which comprises a shell and a loudspeaker, wherein the loudspeaker is arranged in the shell; the shell is made of a magnesium base composite material; the magnesium base composite material comprises magnesium base metal and nano reinforcing phases which are dispersed in the magnesium base metal; and compared with an earphone which has the same shape and adopts a shell made of AZ91D magnesium alloy, the earphone reduces the total harmonic distortion by 10 percent in a frequency range of 20Hz to 50Hz.

The invention provides a loudspeaker array sound reinforcement system generating a uniform sound field, comprising: an input signal processing module (1), a filtering module (2), a multi-channel power amplifier (3) and a loudspeaker array (4); the filtering module (2) comprising: a parameter setting unit (201), a filter bank generating unit (202) and a filter bank processing unit (203); the filter bank generating unit (202) is configured to set The parameter design optimization problem set by unit (201): in the actual sound field energy efficiency is greater than or equal to J c Under the constraint condition, find a vector solution, make the difference of the actual sound field distribution calculated and target sound field distribution minimum; This vector solution is complex number weight coefficient vector; According to described complex number weight coefficient vector, obtain described filter bank processing unit ( 203) the time-domain impulse responses of the L filters. The sound reinforcement system of the invention can form a uniformly distributed sound field in the target area.

In an inner bottom surface of a case, a substantially oblong recess having a long axis and a short axis forms a vibration area. A piezoelectric element is bonded to the center of the recess. On the opposite sides of the vibration area, vibration suppression areas thicker than the vibration area are disposed. A side portion of the case is formed to be thin over the entire circumference thereof. A reinforcing member higher in rigidity than the case is bonded to upper portions of the vibration suppression areas. The reinforcing member has a bottom surface substantially equal to the shape of the vibration suppression areas, and has a predetermined height. A gap between the reinforcing member and an inner side surface of the case is also filled with a filling member.

The invention relates to the field of audio data processing, in particular to a sound reinforcement system and a microphone channel data selection method, which greatly reduce the influence of reverberation and noise, facilitate the expansion of the coverage range and ensure the sound reinforcement quality. The microphone channel data selection method comprises the following steps: a sampling rate and a sampling bit number of data acquisition of each microphone channel are configured; the analog signals collected by the microphones are converted into digital signals according to the configured sampling rate and sampling digits, and the digital signals are sent to the audio processing module; the audio processing module is used for processing the digital signal, selecting corresponding microphone channel data according to a processing result, and sending the corresponding microphone channel data to the audio output module; and the audio output module converts the processed digital signal into an analog signal and sends the analog signal to sound equipment for output. The sound amplification device is suitable for sound amplification of places such as classrooms, meeting rooms and lecture halls.

The invention relates to a sound-absorbing structure of a sound-proof cover, which consists of a galvanized outer shell, a perforated inner wall, glass fiber cloth, and a sound-absorbing material. A sound-absorbing material is arranged between the galvanized outer shell and the perforated inner wall. The surface of the material is covered with a layer of glass fiber cloth. The distance between the galvanized sheet of the outer shell and the perforated sheet of the inner wall is not less than 50mm. The perforated plate on the inner wall is provided with several perforations, and the perforation rate is not less than 20%. The invention has a compact and reasonable structure; it can effectively reduce the reverberation in the sound-proof enclosure and improve the sound absorption frequency band of the perforated plate; the surface of the sound-absorbing material is covered with a layer of glass fiber cloth to prevent fine chips or fibers from being scattered from the perforated plate; it can improve The sound absorption coefficient of the inner wall improves the noise reduction ability; and it has the characteristics of sound insulation and sound absorption effect.

There are provided piezoelectric element, case having conductivity, first acoustic matching layer bonded on case, and second acoustic matching layer stacked on and bonded to first acoustic matching layer. A joining part between first acoustic matching layer and second acoustic matching layer is located inside an outer periphery of a joining surface of first acoustic matching layer that is joined to second acoustic matching layer. This configuration can prevent indirect wave generated in first acoustic matching layer from propagating to second acoustic matching layer, and can therefore reduce reverberation of an ultrasonic wave.

The invention discloses a method and a device for ultrasonic image space compound imaging. The method comprises the following steps of: receiving ultrasonic echo signals with different deflection angles; performing image space compound imaging for the ultrasonic echo signals with different deflection angles; calculating the variance of the image data after the present compound imaging with the image data after the previous compound imaging, and then getting the absolute value, so as to obtain the variance image data, comparing the variance image data with a preset variance threshold, and judging whether the image data with a high motion attribute exists in the image data after the present compound imaging; and determining the output mode of the image data according to whether the image data with a high motion attribute exists in the image data after the present compound imaging. In the technical scheme of the invention, different image processing modes can be chosen according to the judged image motion attribute by judging the image data with a high motion attribute; and speckle noise, sound shadow and reverberation in an ultrasonic image can be effectively reduced, thus improving the definition of the image.

The invention relates to the technical field of park facilities, and discloses a forest park hidden waterproof and dustproof sound box, which comprises a hollow supporting column, the top of the hollow supporting column is fixedly connected with a connecting plate, the top of the connecting plate is fixedly connected with a supporting cylinder, and the top of the supporting cylinder is fixedly connected with a sponge pad. The top of the connecting plate is fixedly connected with a top cover, the sponge pad is in contact with the inner wall of the top of the top cover, a sealing pad is arranged between the top cover and the connecting plate, the circumferential outer wall of the supporting cylinder is sleeved with a connecting ring, and the bottom of the connecting ring is fixedly connected with a sound box body. The sealing performance of the connecting position of the top cover and the connecting plate is improved through a sealing gasket, so that internal seepage of rainwater is avoided, then the bottom end of the hollow supporting column is sealed through a sealing plug and a bottom pad, accumulated water is prevented from entering the hollow supporting column from one end of the hollow supporting column, damage to electric wires in the hollow supporting column is avoided, and normal use of the sound box in a forest park is ensured.

A pain relief device and method or system to aid in the resolution of pain in a body including at least one layer of PVDF film and at least one other layer for directional purposes, as well as a polarizing layer. The device can be packaged in various ways.