121 results about "Octave band" patented technology

A multistage equalizer filter (100) is presented for compensating a received distorted signal having an amplitude for frequency dependent signal modifications introduced by a transmission channel. The filter (100) comprises at least one amplifying compensation stage (21a, 21b, 21c, 21d) having a gain and a saturation level, the gain being monotonically rising for at least a last decade in frequency below an upper data frequency of the received signal, and a gain control means (125) for controlling the gain of the amplifying compensation stage (21a, 21b, 21c, 21d) such that the amplitude of the received signal amplified in the at least one amplifying compensation stage (21a, 21b, 21c, 21d) remains below the saturation level of the amplifying compensation stage (21a, 21b, 21c, 21d). The gain may be regulated by separately regulating low frequency gain, mid-frequency gain and higher frequency gain for each of the at least one amplifying compensation stages (21a, 21b, 21c, 21d).

The invention concerns an acoustic sensor (44) for identifying obstruction of a circular grit trap for recycling fine sand in an industrial water clarifying plant. The circular grit trap comprises a cylindrical body having an outer wall and having at one first end a first sand-containing fluid inlet for receiving sludge and fine sand and a first overflow outlet of fluid not containing sand orthogonal to the fluid inlet, to evacuate said sludge, and at a second end a second sand-containing fluid underflow outlet coaxial with the first fluid outlet, to recover said sand. The acoustic sensor is sensitive to the noise radiated by the flow of the sand-containing fluid in the circular grit trap and is applied against the outer wall of the cylindrical body of the circular grit trap, in the plane of the fluid inlet but not coaxially therewith. Said acoustic sensor transmits a warning signal when an abnormal amplitude variation of sound level is measured in the bands of ⅓ octave centered on frequencies of 25 Hz or 200 Hz.

Low-frequency duct noise is relatively difficult to deal with by passive methods such as porous duct lining. Reactive methods like expansion chamber are rather bulky, while compact resonators are to narrow-banded. This invention shows that a suitably stetched thin membrane backed by a slender cavity can achieve a satisfactory performance from low to medium frequencies over an octave band.

Data signals transmitted over transmission media suffer from attenuation caused by the transmission media. Equalization circuitry may be provided to compensate for attenuation caused by the transmission media. Equalization circuitry may include multiple stages arranged in series to allow the frequency responses of the stages to aggregate together. Each stage may be programmable to insert a zero, which causes the frequency response of the stage to increase in magnitude by 20 dB / decade. The frequency location of the zero may also be programmable to allow each stage to contribute a certain amount of gain for a specific frequency. Each stage may also be programmable to determine the location of poles for reduction of high frequency noise and cross-talk cancellation.

A method and apparatus to compensate for imperfections in a sound field includes determining a frequency-response characteristic of a test signal, extracting a difference spectrum by filtering the frequency-response characteristic on different octave band scales, detecting frequency parameters of peak and dip components that deviate from a critical value in the difference spectrum, and calculating filter coefficients from the frequency parameters to filter an input audio signal.

Waveforms for controlled source electromagnetic surveying. The waveforms have frequency spectra that include three or more frequencies spaced at substantially equal intervals on a logarithmic frequency scale and spanning a bandwidth of about one decade or more, at least three of which frequencies have approximately equal corresponding amplitudes.

Adaptive entropy encoding and decoding which utilizes Set Partitioning within Generalized Hierarchical Trees (SPRIGHT) and a method of designing trees utilizing directionality. After decorrelation and quantization a tree structure is selected from multiple candidates, based on geometric relationships within the image block, for coding the coefficients toward improving zero-clustering of coefficients. Trees for the SPRIGHT encoding are created in response to finding frequency position of each coefficient and scaling frequency position followed by use of octave-band partitioning of coefficient patterns into squares and L-shapes, and the L-shapes are iteratively partitioned into squares. The tree comprises leaf nodes containing coefficients associated with each non-leaf node. The number of zero clustered coefficients can be increased, thus decreasing the number of nodes coded into the encoded image output.

A unique, fully integrated, fully programmable, and highly flexible sound distribution system and methodology for providing masking sound, background music, and paging capabilities in up to eight zones of a building or space is provided. The methodology embodied in the system includes internal masking sounds that are uniquely pre-filtered to provide efficient and effective masking of distracting sounds within selectable zones of the space with a minimum masking sound dB sound level and with a pleasant sounding and non-annoying masking sound. The system also incorporates the capacity to be controlled from a remote or local telephone to adjust the volume level in any zone serviced by the system by issuing appropriate DTMF codes from the telephone's keypad. Unique bi-tone diagnostic functions are provided for assuring that the entire system is correctly wired and installed and for troubleshooting operational anomalies. ⅓ octave equalization is provided to compensate for known frequency response characteristics of the flat panel radiators of the system and to compensate for varying room acoustics to provide a low special variation of sound among the various zones of the space. The result is a high quality high fidelity sound that is consistent from zone to zone.

A film comprising from about 20 wt. % to about 85 wt. % of a polyolefin component, from about 5 wt. % to about 30 wt. % of an elastomeric resin, and from about 20 wt. % to about 45 wt. % of a particle component; wherein the film has a basis weight of from about 5 gsm to about 25 gsm; wherein the film has a predicted sound pressure level over the frequency octave range of 2000 Hz-6300 Hz of less than about 43 dB.

A phase-linear wide band converter for frequency conversion from a multi-octave baseband (BB) to an RF band or vice versa and utilizing I / Q signal processing. The converter uses a phase-linear power divider / combiner having three ports, an I / Q mixer stage, an I / Q power divider for an I / Q local oscillator (LO) signal, two wide band balanced mixers, and impedance matching circuits. The improved performance over multi-octave bands is due to the mutually adaptation between an active opamp I / Q power divider / combiner at an extended baseband, a single low-pass filter (LPF) having predetermined amplitude and phase characteristics and being inserted in the baseband signal path to or from the active baseband power divider / combiner, and a baseband phase correction stage.

In some embodiments, a system may include a passive uniplanar single-balanced millimeter-wave mixer. In some embodiments, a three-port diode-tee IC forming a mixer core is coupled between an end of a slotline balun and a second coplanar balun. The operational bandwidth of a mixer structure is enhanced by optimizing the distance between the mixer diode-tee core and the back-short circuits. The frequency separation of LO and IF signals may be accomplished by means of stand-alone three-port filter-diplexer device. The system may allow wider than a frequency octave operational bandwidth for a frequency converter device all the way into millimeter wave frequencies at the same time as supporting the operational bandwidth for baseband IF signal over more than six frequency octaves. In some embodiments, the system may accomplish a 500 MHz to 34.5 GHz continuous IF bandwidth with RF signal sweeping from 33 GHz to 67 GHz and local oscillator at 67.5 GHz fixed frequency.

The invention discloses a high-temperature-condition sound field measurement and analysis device and an analysis method thereof. The device comprises a sound guide tube array, a flexible semi-infinite tube, a sound sensor, a data collector and a computer. Through the combination design of the sound guide tubes and the semi-infinite tube, the tail end reflection of the guide tube can be restrained, and the measurement accuracy can be enhanced. The device can conduct sound field data acquisition and processing, and can realize the time domain, frequency domain and reconstruction analysis of a sound field. In the data processing process, data at the tail end of the sound guide tube can be corrected by utilizing a method of correcting the sound pressure amplitude value of the one-third octave band-based sound guide tube, and the measurement accuracy can be enhanced; the optimization of a conventional beam-formed sound field reconstruction method can be realized through kurtosis weighing computation, and accurate sound pressure distribution and strength information can be obtained. The high-temperature-condition sound field measurement and analysis device and the analysis method thereof have the advantages of being capable of conducting the sound field measurement under severe environments in which the conventional microphone can not directly conduct measurement, accurately reconstructing the distribution and strength information of the sound field at a high-temperature environment, and providing a data support for the fatigue invalidation analysis of materials / members under thermal sound coupling environments.

Waveforms for controlled source electromagnetic surveying. The waveforms have frequency spectra that include three or more frequencies spaced at substantially equal intervals on a logarithmic frequency scale and spanning a bandwidth of about one decade or more, at least three of which frequencies have approximately equal corresponding amplitudes.

The invention discloses a noise and vibration measurement system based on a virtual instrument technology, which is equipped with analogue signal input and analogue signal output. Signals can be input and output simultaneously or separately. An analogue signal can be input by two modes, i.e. IEPE sensor input and voltage signal input; and the output mode thereof is analogue voltage signal output. The input and the output are both connected with a data acquisition module through BNC wires, the data acquisition module is connected with a computer through a USB to input a sensor signal to the computer; and a computer software processing module integrates the following analysis and measurement modules: a spectrum analysis module, an octave band analysis module, a transfer function measurement module and a recorded signal viewing module.

Low-frequency duct noise is relatively difficult to deli with by passive methods such as porous duct lining. Reactive methods like expansion chamber are rather bulky, while compact resonators are too narrow-banded. This invention shows that a suitably stretched thin membrane backed by a slender cavity can achieve a satisfactory performance from low to medium frequencies over an octave band.

The invention discloses a full-digital real-time noise measurement and analysis method and a system thereof. The noise measurement and analysis system provided with a noise measurement and analysis software PDA is connected with a noise collection system by a USB port. The method and system realize the digitization of frequency weighting, time weighting, octave and 1 / 3 octave processing, and can display time domain waveform and frequency domain waveform and provide system settings; and a user can select settings of sampling frequency and sampling time of the noise collection system, select L linearity, A and C types of frequency weighting and select time average, fast gear and slow gear of time weighting. In a graphic display mode, the user selects a frequency domain analysis method of signals and displays a corresponding frequency spectrogram according to needs, wherein the functions of FFT frequency spectrogram analysis, octave frequency spectrum analysis, 1 / 3 octave frequency spectrogram analysis and the like realize the measurement and analysis to real-time noise. The method and system utilize a PDA computer system to realize the full-digital real-time noise signal measurement and analysis with low cost and high quality.

Provided is a switched multiplexer configured to combine first, second, third and fourth signal paths each covering a sub-octave in a frequency range of from about 2 to about 18 GHz and to remove transmitter harmonics of an input signal to create a stable output impedance across the frequency range. The switched multiplexer comprises a transmit switch, a first diplexer and a power combiner. The first diplexer is connected in parallel with a second diplexer. The first diplexer comprises first and second signal paths. The second diplexer comprises third and fourth signal paths. Each one of the first, second and third signal paths include respective ones of the first switch, a second switch and a third switch, each interconnected to respective ones of a first signal path high-pass filter, second signal path low-pass filter and third signal path low-pass filter by respective ones of a first, second and third filter. Each one of the first, second and third filters have a respective filter capability in the range of from about 6-10.4 GHz, 2-3.5 GHz, and 3.5-6 GHz.

The invention provides a power transformer fault prediction and diagnosis method based on audio characteristics. The method specifically comprises the steps that S1, detecting effective signals of power transformer audio data under the noise background based on chaotic oscillators; S2, calculating a logarithm energy spectrum on the nonlinear Mel scale to serve as a characteristic quantity of the audio signal of a power transformer; S3, calculating a principal component of the audio signal characteristic quantity of the power transformer by adopting a principal component analysis method; S4, optimizing the optimal hyper-parameter training power transformer fault prediction model of the vector machine algorithm by adopting a quantum particle swarm optimization algorithm; and S5, if the powertransformer is in a fault state, adopting a 1 / 3 octave algorithm to extract the fault characteristic frequency range amplitude, comparing the fault characteristic frequency range amplitude with an expert experience rule base, and predicting / obtaining the fault type of the power transformer. According to the method, the identification precision of power transformer operation state fault predictioncan be improved, and the calculation amount is reduced; fault judgment can be carried out based on other measurement data or infrared images.

A method to test multichannel noise and shake based on computer processing has two signal input mode: hardware input and file input used for signal ex post analysis. In hardware input, firstly, take sensor and data collecting module to form continuous data collecting module connected by BNC. Then, connect the module with computer by USB for inputting signal into computer. In file input, program read data from designated file and analyze. Its analyzing process is: set parameters for task, which includes signal analyzing parameters, window function selection of spectrum and coherence analysis, average mode and its parameters, selection of average mode and weighted window function in octave-band analysis. If in hardware input, it needs point file position.

The invention discloses a digitized acoustic detection system, which comprises a microphone, a front end circuit, an analog-to-digital (A / D) converter, a signal processor, a touch screen and a keyboard. After the microphone converts an acoustic signal into an electric signal, the signal is input into the front end circuit. After the front end circuit amplifies the signal, the electric signal is input into the A / D converter for analog-to-digital conversion. The converted signal is input into the signal processor for signal processing. The touch screen and the keyboard are connected with the signal processor. A calibration module, a sound level measuring module, a spectrum analysis module and a fast Fourier transform (FFT) analysis module are arranged in the signal processor, wherein the calibration module is used for channel calibration and microphone calibration; the sound level measuring module supports A, C and Z frequency weighting; the spectrum analysis module is used for conducting real-time measurement by using 1 / 1 and 1 / 3 octave frequency bands within an allowable wide frequency range; and the FFT analysis module is used for measuring the transient sound level, the maximum sound level and the minimum sound level of the selected frequency range. The digitized acoustic detection system has the advantages that the basic detection of various sound levels and sound intensity calculation can be realized, the real-time spectrum and statistical analysis can be conducted to sound vibration signals, the sound vibration signals can be automatically monitored, the data redundancy of low-frequency parts is reduced and the spectrum leakage is reduced through windowing.

The invention relates to a measuring and estimating method for civil architecture room noise in the field of environmental noise assessment, which comprises the following steps: A. choosing measuring point and measuring A medium of sound level (LA), getting real value of octave-band sound pressure level at the same time; B. computing the value of NR by using equation NR= LA-b and the measured A medium of sound level (LA); C. corresponding the value of step B and the NR curve tablet as normalized value of octave-band sound pressure level; D. comparing the real value of octave-band sound pressure level from A and normalized value from C.

A DSP subtracts the sound pressure Pgen of a notification sound from the sound pressure Pout of an octave band including the notification sound among the sound pressures P1 to P8 of octave bands having center frequencies f1 to f8 acquired by the octave analysis of ambient noises including the notification sound, and thereby acquires the octave band characteristics of only the ambient noises. Next, the DSP detects the sound pressure Pmin of an octave band of the ambient noises which has the lowest sound pressure and the sound pressure Pmax of an octave band of the ambient noises which has the highest sound pressure from these octave band characteristics. Then, the DSP generates an output waveform of the sound pressure Pgen that is almost equal to the sound pressure Pmax of the octave band having the highest sound pressure in the octave band having the lowest sound pressure.

An aspect of the present invention is a voltage controlled oscillator that provides multi octave band tenability. The oscillator preferably includes one or more multi planar resonators which allow it to be implemented in integrated circuit form. Oscillators implemented in accordance with the aspect of the present invention provide as wide a tuning range as the YIG based resonators but are much more compact and efficient. Therefore, they are amenable to implementation in integrated circuits and suited for adaptation in third and future generation wireless devices.

The invention provides a gain calculation method for a caustic convergence region under a deep sea complete sound channel based on a ray normal mode theory. The gain calculation method is characterized in that under the deep sea complete sound channel condition, a typical deep sea Munk sound velocity profile situation is analyzed by using a ray normal mode theory, and then an expression of the sound pressure is obtained; the sound intensity is calculated according to the expression of the sound pressure; the average sound intensity value of the energy is calculated in the bandwidth of the 1 / 3octave of the central frequency; the average propagation loss value of the energy is calculated; the spherical wave propagation loss value at the caustics of the convergence region is calculated; andthe energy average propagation loss value is compared with the spherical wave propagation loss value, so that the gain value at the caustic convergence region is solved, and the result proves that thegain calculation method provided by the invention has a very good effect.

The invention discloses a pump station unit fault diagnosis method based on noise signal A-weighted analysis, and relates to the technical field of pump station unit fault diagnosis. The method comprises the following steps: monitoring a noise sound pressure signal of a pump station unit, performing spectral analysis on the sound pressure signal, performing A-weighted network correction on a sound pressure level of each frequency component, performing energy superposition on the sound pressure levels to obtain an A-weighted noise sound pressure level, and when the sound pressure level of the A-weighted noise exceeds an alarm value, performing 1 / 3 octave spectrum analysis on the sound pressure signal to extract energy characteristics of each octave frequency band. A deep extreme learning machine is utilized to quickly and effectively perform characteristic learning and extract implicit fault information of each characteristic, so that intelligent diagnosis of faults of the water pump unit is realized.

The invention provides a method for determining the noise equivalent frequency of a motor vehicle, which is suitable for different travel speeds and different vehicle types, has simple calculation and high precision, and is used for accurately calculating the insertion loss of a sound barrier. The method for determining the noise equivalent frequency of the motor vehicle comprises the following steps of: step 1, acquiring the noise sample data of the vehicle, including 1 / 3 octave sound pressure level of a single vehicle, the vehicle type and the travel speed; step 2, using a method of Appendix B of Norm on Acoustical Design and Measurement of Noise Barriers to calculate the noise equivalent frequency; and step 3, analyzing a great number of sample data in a statistical manner according to different vehicle type intervals and different travel speed intervals, and determining the noise equivalent frequency which has the maximum probability of occurrence.

A system and method for enabling multiple sub-octave band transmissions with reduced second order distortions is provided. For this method, first and second sub-octave bands are established. The second sub-octave band is spaced from the first sub-octave band by a non-transmission band. Digital signals are modulated onto RF carrier frequencies in the first and second band to produce first band RF signals and second band RF signals. The first and second band signals are converted into one or more light beams and transmitted over a fiber optic cable. After transmission, an optical receiver reconverts the light beam into an RF signal. Second order distortions outside a selected sub-octave band can be filtered from RF signal and a tuner used to tune in a selected carrier frequency. A receive modem can then be used to demodulate the tuned carrier frequency for receipt of its respective digital signal.

The invention discloses a method for improving indoor speech articulation. The method comprises steps as follows: geometrical size of an interior building is acquired, and an indoor simulation model is established according to the geometrical size; a plurality of parameter evaluation points are selected from the indoor simulation model, and a plurality of articulation weak points are selected fromthe parameter evaluation points according to a preset articulation evaluation parameter calculation strategy; the articulation weak points are used as reference positions for indoor layout of diffusers and reflection materials; average reverberation time of the articulation weak points at an octave frequency band is calculated; sound absorption increment corresponding to the octave frequency bandis calculated according to preset target reverberation time of the octave frequency band and the average reverberation time; a sound absorption material is selected according to the sound absorptionincrement; and the sound absorption material is used for being arranged in a first preset position of the interior. The invention further discloses a system for improving the indoor speech articulation. With adoption of the embodiment, the indoor speech articulation can be effectively improved.