700 results about "Frequency resolution" patented technology

Carrier Interferometry (CI) provides wideband transmission protocols with frequency-band selectivity to improve interference rejection, reduce multipath fading, and enable operation across non-continuous frequency bands. Direct-sequence protocols, such as DS-CDMA, are provided with CI to greatly improve performance and reduce transceiver complexity. CI introduces families of orthogonal polyphase codes that can be used for channel coding, spreading, and / or multiple access. Unlike conventional DS-CDMA, CI coding is not necessary for energy spreading because a set of CI carriers has an inherently wide aggregate bandwidth. Instead, CI codes are used for channelization, energy smoothing in the frequency domain, and interference suppression. CI-based ultra-wideband protocols are implemented via frequency-domain processing to reduce synchronization problems, transceiver complexity, and poor multipath performance of conventional ultra-wideband systems. CI allows wideband protocols to be implemented with space-frequency processing and other array-processing techniques to provide either or both diversity combining and sub-space processing. CI also enables spatial processing without antenna arrays. Even the bandwidth efficiency of multicarrier protocols is greatly enhanced with CI. CI-based wavelets avoid time and frequency resolution trade-offs associated with conventional wavelet processing. CI-based Fourier transforms eliminate all multiplications, which greatly simplifies multi-frequency processing. The quantum-wave principles of CI improve all types of baseband and radio processing.

Carrier Interferometry (CI) provides wideband transmission protocols with frequency-band selectivity to improve interference rejection, reduce multipath fading, and enable operation across non-continuous frequency bands. Direct-sequence protocols, such as DS-CDMA, are provided with CI to greatly improve performance and reduce transceiver complexity. CI introduces families of orthogonal polyphase codes that can be used for channel coding, spreading, and / or multiple access. Unlike conventional DS-CDMA, CI coding is not necessary for energy spreading because a set of CI carriers has an inherently wide aggregate bandwidth. Instead, CI codes are used for channelization, energy smoothing in the frequency domain, and interference suppression. CI-based ultra-wideband protocols are implemented via frequency-domain processing to reduce synchronization problems, transceiver complexity, and poor multipath performance of conventional ultra-wideband systems. CI allows wideband protocols to be implemented with space-frequency processing and other array-processing techniques to provide either or both diversity combining and sub-space processing. CI also enables spatial processing without antenna arrays. Even the bandwidth efficiency of multicarrier protocols is greatly enhanced with CI. CI-based wavelets avoid time and frequency resolution trade-offs associated with conventional wavelet processing. CI-based Fourier transforms eliminate all multiplications, which greatly simplifies multi-frequency processing. The quantum-wave principles of CI improve all types of baseband and radio processing.

An electronic scanning type radar device mounted on a moving body includes: a transmission unit transmitting a transmission wave; a reception unit comprising a plurality of antennas receiving a reflection wave of the transmission wave from a target; a beat signal generation unit generating a beat signal from the transmission wave and the reflection wave; a frequency resolution processing unit frequency computing a complex number data; a target detection unit detecting an existence of the target; a correlation matrix computation unit computing a correlation matrix from each of a complex number data of a detected beat frequency; a target consolidation processing unit linking the target in a present detection cycle and a past detection cycle; a correlation matrix filtering unit generating an averaged correlation matrix by weighted averaging a correlation matrix of a target in the present detection cycle and a correlation matrix of a related target in the past detection cycle; and a direction detection unit computing an arrival direction of the reflection wave based on the averaged correlation matrix.

An audio encoder for encoding an audio signal has a first coding branch, the first coding branch comprising a first converter for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder has a second coding branch comprising a second time / frequency converter. Additionally, a signal analyzer for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time / frequency resolution to be applied by the converters when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time / frequency converter and used by the second time / frequency converter.

An audio decoder for decoding a multi-audio-object signal having an audio signal of a first type and an audio signal of a second type encoded therein is described, the multi-audio-object signal having a downmix signal and side information, the side information having level information of the audio signals of the first and second types in a first predetermined time / frequency resolution, and a residual signal specifying residual level values in a second predetermined time / frequency resolution, the audio decoder having a processor for computing prediction coefficients based on the level information; and an up-mixer for up-mixing the downmix signal based on the prediction coefficients and the residual signal to obtain a first up-mix audio signal approximating the audio signal of the first type and / or a second up-mix audio signal approximating the audio signal of the second type.

A method for decoding a multi-audio-object signal having audio signals of first and second types encoded therein, the multi-audio-object signal having a downmix signal and side information having level information of the audio signals of the first and second types in a first predetermined time / frequency resolution, the method including computing a prediction coefficient matrix C based on the level information; and up-mixing the downmix signal based on the prediction coefficients to obtain a first and / or a second up-mix audio signal approximating the audio signals of the first and second types, respectively, wherein up-mixing yields the first and / or second up-mix signals S1 and S2 from the downmix signal d according to a computation representable by(S1S2)=D-1{(1C)d+H},with “1” denoting—depending on the number of channels of d—a scalar, or an identity matrix, and D−1 being a matrix uniquely determined by a downmix prescription according to which the audio signals of the first and second types are downmixed into the downmix signal, and which is also included by the side information, and H being a term independent from d.

A low noise multi-loop radio frequency synthesizer is disclosed for the read channel in a hard disk drive, and for RF wireless communications local oscillator applications. The frequency synthesizer receives an input reference signal having a frequency fR, into a fine tune phase locked loop and into a coarse tune phase locked loop. Driven by the input reference signal, the fine tune PLL outputs a fine tune signal with a frequency fR.P, where P is an integer, while the coarse tune PLL, also driven by the same input reference signal, outputs a coarse tune signal with a frequency fR.A, where A is an integer. A translation phase locked loop has a unity multiplication factor and is driven by the fine tune signal output. The frequency synthesizer finally has a Gilbert cell double balanced mixer coupled between the coarse tune and the translation phase locked loops, wherein the Gilbert cell mixer combines the coarse tune signal and the output signal of the translation phase locked loop and couples the mixed signal into the translation phase locked loop. The translation loop outputs a signal with a frequency which is proportional to the linear sum of the coarse tune signal and the fine tune signal.

An enhancement system improves the estimate of noise from a received signal. The system includes a spectrum monitor that divides a portion of the signal at more than one frequency resolution. Adaptation logic derives a noise adaptation factor of the received signal. A plurality of devices tracks the characteristics of an estimated noise in the received signal and modifies multiple noise adaptation rates. Weighting logic applies the modified noise adaptation rates derived from the signal divided at a first frequency resolution to the signal divided at a second frequency resolution.

An electronic scanning type radar device mounted on a moving body includes: a transmission unit transmitting a transmission wave; a reception unit comprising a plurality of antennas receiving a reflection wave of the transmission wave from a target; a beat signal generation unit generating a beat signal from the transmission wave and the reflection wave; a frequency resolution processing unit frequency computing a complex number data; a target detection unit detecting an existence of the target; a correlation matrix computation unit computing a correlation matrix from each of a complex number data of a detected beat frequency; a target consolidation processing unit linking the target in a present detection cycle and a past detection cycle; a correlation matrix filtering unit generating an averaged correlation matrix by weighted averaging a correlation matrix of a target in the present detection cycle and a correlation matrix of a related target in the past detection cycle; and a direction detection unit computing an arrival direction of the reflection wave based on the averaged correlation matrix.

The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular; a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described, The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of Δf, The system further comprises a nonlinear processing unit (502) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P; and a synthesis filter bank (504) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank (504) has a frequency resolution of FΔf; with F being a resolution factor, with F≧1; wherein the transposition order P is different from the resolution factor F.

An audio encoder for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder including a processor for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder further includes an entropy encoder for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream.

The invention provides a micro-gesture identification method and system based on ultrasonic active detection. The method includes the following steps of emitting an ultrasonic signal and receiving an ultrasonic echo signal, reflected by an identified gesture action, of the ultrasonic signal; comparing the ultrasonic signal with the ultrasonic echo signal to obtain the reception time delay and frequency resolution variance of the ultrasonic signal and the ultrasonic echo signal; according to the reception time delay and frequency resolution variance, obtaining the distance and velocity information of the timing sequence in the ultrasonic echo signal; and according to the distance and velocity information, performing identification and determination on the identified gesture action through a data driving pattern recognition algorithm. The micro-gesture identification method and system can accurately identify the micro-gesture actions and have the advantages of high resolution, high robustness, low consumption, and low algorithm complexity.

A method and device for automatically assessing loss of hearing sensitivity and compression (recruitment) with user defined frequency resolution by means of extrapolated DPOAE I / O functions and ABRs as well as for automatically fitting hearing aids without any cooperation of the subject tested using a device having a display screen attached to a handheld device generating and collecting otoacoustic emission signals and brain stem response signals into a programmed with a clinical audiogram with fitting parameters for hearing aids calculated on the basis of assessed hearing threshold and compression and identifying the type of hearing required for the individual.

A serial bus clock frequency calibration system and a method thereof are disclosed herein. The system utilizes a first frequency calibration device, a second frequency calibration device and a third frequency calibration device to share the same oscillator as so to perform multi-stage clock frequency resolution calibrations for different frequency-tuning ranges. This can bring an optimal frequency resolution, greatly reduce system complexity and save element cost.

In a frequency analysis system, such as a signal detection system or a spectrum analyzer, the frequency domain resolution is enhanced by compression and decompression of the signal samples. The limited capacity of the data storage and / or data transfer resources limit the number of samples that can be stored or transferred. A compressor forms a compressed signal prior to data transfer or storage. A decompressor decompresses the compressed signal prior to transformation to the frequency domain, by a fast Fourier transform or other frequency domain transform. The frequency domain resolution is enhanced because more decompressed samples are available for the frequency domain transform. The compressor and decompressor apply computationally efficient algorithms that can be implemented to operate in real time.

A low noise multi-loop radio frequency synthesizer receives an input reference signal having a frequency fR, into a fine tune PLL and a coarse tune PLL. The fine tune PLL outputs a fine tune signal with a frequency fR□P, P beings an integer, while the coarse tune PLL outputs a coarse tune signal with frequency fR□A, where A is an integer. A translation PLL has a unity multiplication factor and is driven by the fine tune signal output. The frequency synthesizer has a Gilbert cell double balanced mixer coupled between the coarse tune and the translation PLLs, the Gilbert cell mixer combining the coarse tune signal and the output signal of the translation PLL and coupling the mixed signal into the translation PLL. The translation loop outputs a signal with a frequency proportional to the linear sum of the coarse tune signal and the fine tune signal.

An inversion-based algorithm for computing the time frequency analysis of reflection seismograms using constrained least-squares spectral analysis is formulated and applied to modeled seismic waveforms and real seismic data. The Fourier series coefficients are computed as a function of time directly by inverting a basis of truncated sinusoidal kernels for a moving time window. Spectra may be provided that have reduced window smearing for a given window length relative to the discrete Fourier transform irrespective of window shape, and a time-frequency analysis with a combination of time and frequency resolution that is superior to the short time Fourier transform and the continuous wavelet transform. The reduction in spectral smoothing enables enhanced determination of spectral characteristics of interfering reflections within a short window. The degree of resolution improvement relative to the short time Fourier transform increases as window length decreases.

An automatic frequency calibration apparatus and a method thereof for a phase-locked loop based frequency synthesizer are disclosed. The apparatus includes a frequency-to-digital converter configured to convert a frequency of a VCO output signal to a first digital value, a target value setting section configured to provide a second digital value corresponding to a target frequency, and a finite state machine configured to calibrate the frequency of the VCO output signal by using the difference of the first digital value and the second digital value. Accordingly, the calibration speed and a frequency resolution of the automatic frequency calibration apparatus in a frequency synthesizer may be enhanced.

A serial bus clock frequency calibration system and a method thereof are disclosed herein. The system utilizes a first frequency calibration device and a second frequency calibration device both to share an oscillator as so to perform two-stage clock frequency resolution calibrations for generating different frequency-tuning ranges. This can bring an optimal frequency resolution and greatly reduce system complexity and save element cost.

An audio encoder for encoding an audio signal comprises a first coding branch (400), the first coding branch comprising a first converter (410) for converting a signal from a time domain into a frequency domain. Furthermore, the audio encoder comprises a second coding branch (500) comprising a second time / frequency converter (523). Additionally, a signal analyzer (300 / 525) for analyzing the audio signal is provided. The signal analyzer, on the hand, determines whether an audio portion is effective in the encoder output signal as a first encoded signal from the first encoding branch or as a second encoded signal from a second encoding branch. On the other hand, the signal analyzer determines a time / frequency resolution to be applied by the converters (410, 523) when generating the encoded signals. An output interface includes, in addition to the first encoded signal and the second encoded signal, a resolution information identifying the resolution used by the first time / frequency converter and used by the second time / frequency converter.

A method and device for automatically assessing loss of hearing sensitivity and compression (recruitment) with user defined frequency resolution by means of extrapolated DPOAE I / O functions and ABRs as well as for automatically fitting hearing aids without any cooperation of the subject tested using a device having a display screen attached to a handheld device generating and collecting otoacoustic emission signals and brain stem response signals into a programmed with a clinical audiogram with fitting parameters for hearing aids calculated on the basis of assessed hearing threshold and compression and identifying the type of hearing required for the individual.

An inductance-capacitance numerical control oscillator in the communication technical field comprises a control deposit unit, a coupling MOS tube, an integrated inductance and a plurality of capacitance arrays, wherein, an input end of the control deposit unit receives an array tuning word output by a numerical control bus, an output end of the control deposit unit is connected with a plurality ofthe capacitance arrays in parallel, and the coupling MOS tube and the integrated inductance are respectively connected with the capacitance arrays in parallel. Compared with the prior art, output frequency resolution ratio is 10 times higher in the case of no capacitance shaking; and if applied in an all digital phase-locked loop, the high frequency resolution ratio of the numerical control oscillator means the integrally output low-phase noise. In addition, the numerical control oscillator of the invention can eliminate the capacitance arrays using delta-sigma modulator shaking control in abetter process (characteristic dimension is less than 65nm) or some communication standards, thus realizing simplifying circuit design further and reducing power consumption and occupied area.

The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency QΩ+Ωq is generated on the basis of existing components at Ω and QΩ+Ωq. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and / or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank (101), a subband processing unit (102) configurable by control data (104) and a synthesis filter bank (103).

A method and device to track navigational satellite signals, are claimed. In this invention, a combination of down-sampling and frequency domain transformation are used to track the navigational satellite signals under dynamic environment. A Fast Fourier Transform (FFT) with long coherent integration has been employed to determine the varying frequency components with high resolution. By representing a number of correlation values with their average value, it is possible to represent a long sequence of input values by a smaller number of values and a relatively short length FFT can reveal the low frequency components that are present in the signal during tracking operation. A large reduction in the computational load may be achieved using this down-sampling method without compromising on the frequency resolution.

The invention relates to a novel high precision frequency and amplitude measuring method for steady-state harmonic and / or inter-harmonic component in power system. The method of measuring the power system steady-state harmonic and / or inter-harmonic component comprises two steps: firstly, the number and the approximate frequency of the steady-state harmonic and / or inter-harmonic component which can possibly exist in the voltage waveform and current waveform of the power system are primarily measured through the least square method by using the resolution at 1Hz; and then multiband pass filtering is performed on each signal component to accurately judge the number of the steady-state harmonic and / or inter-harmonic component and to measure the frequency and amplitude of each component. The measuring method can swiftly and accurately measure the frequency and amplitude parameter of the steady-state harmonic and / or inter-harmonic component in the voltage waveform and the current waveform of the power system while not be affected by the frequency fluctuation of the power system(namely, nonsynchronous sampling ), and can improve the efficiency of measuring the steady-state harmonic and / or inter-harmonic and reduce the amount of calculation.

The invention provides a phase difference measuring method based on improved windowing discrete Fourier transform. The phase difference measuring method includes the first step of collecting two periodic signals to be measured, the second step of analyzing the collected periodic signals to digital signals, the third step of carrying out windowing processing on the digital signals by constructing a 4-step Blackman-Harris window, and carrying out FFT spectral analysis on the signals subjected to windowing processing to obtain signal frequency spectrums, extracting fundamental wave parameters and respectively calculating initial phase angles of the periodic signals to be measured, and the fourth step of correcting phase positions of useful frequency spectrums according to a discrete spectrum correction method so as to calculate the phase differences of the periodic signals to be measured. The phase difference measuring method can effectively solve the problems that a time domain of a DFT algorithm is cut off, the introduced spectrums are revealed, and spectral analysis has bigger errors caused by the picket fence effect, overcomes the defects of spectrum leakage and lower frequency resolution in an interpolation method, improves the frequency resolution, and finally achieves the high-precision measurement of the phase differences.

In a method of acoustic feedback suppression, digitized time-domain samples of acoustic signals are obtained, and discrete time-frequency transformation is performed on the digitized time-domain samples to generate a plurality of frequency bins of a frequency resolution. A howling frequency bin is identified, where the howling frequency has the maximum magnitude among the plurality of frequency bins. A peak frequency within the howling frequency bin is detected, and this peak frequency is suppressed.

The invention discloses a method used for on-line measurement of density of alkali metal in atom magnetometer air chambers at SERF (spin-exchange relaxation free) states. According to the method, an atom magnetometer is provided with a weak background magnetic field and low frequency stage horizontal linear frequency modulated signals, a data collecting card is used for collecting response signals of the atom magnetometer in frequency sweep time; fast Fourier transform is adopted so as to obtain a magnetic resonance curve in a frequency domain, the resonance line width and the resonant frequency of the magnetic resonance curve are obtained via calculation; the background magnetic field is changed so as to obtain a plurality of resonance line widths and resonant frequencies, quadratic curve fitting of the resonance line widths and the resonant frequencies is carried out so as to obtain spin exchange time; and at last the density of alkali metal in atom magnetometer air chambers at the current temperature can be obtained via calculation. The method can be performed at maintained SERF states, on-line measurement of the density of alkali metal in the air chambers can be realized only using the SERF magnetometer; excellent frequency resolution ratio of linear frequency signals can be achieved at low frequency range (1kHz or lower); and the method is suitable for low frequency scanning range required by SERF magnetometer weak magnetic field.

The invention discloses a method for measuring electric network voltage flicker, and discloses a high-accuracy rapid flicker calculating method aiming at a non-integral cycle low sampling rate by taking account of errors probably caused by harmonic waves and electric network frequency. The core of the calculating method is that: a low fixed sampling rate is adopted to perform non-integral cycle sampling on electric network voltage waveforms, a cosine window is used for performing a fast Fourier transform algorithm (FFT), then the flicker frequency component is accurately extracted through a double-spectral line interpolation algorithm and a frequency spectrum searching algorithm, and the instantaneous flicker visual sensitivity is obtained through calculation. By using the method, under the condition of non-integral cycle low sampling rate, the errors caused by frequency spectrum leakage and harmonic wave aliasing are overcome, the frequency resolution is improved, the complexity of flicker calculation is effectively simplified, and a flicker measurement result is acquired quickly. By using the method, the performance requirement of hardware of an analog-digital converter, a processor and the like is reduced, thus the cost of a flicker measuring instrument is reduced, and simultaneously the measuring accuracy is improved; and besides, the method is applicable to fast low-cost measurement of the electric network voltage flicker, and has a very good effect.

A method for individually fitting a hearing instrument to a user, comprising at least one microphone for generating an input audio signal from ambient sound, an audio signal processing unit for processing the input audio signal into a processed output audio signal, and a transducer for stimulation of the human auditory system according to the processed output audio signal as input to said transducer is described, the method comprising: providing the user with the hearing instrument and starting operation of the hearing instrument; pre-defining a desired target loudness function, wherein loudness perception of a stimulus by the user when using the hearing instrument is defined as function of frequency and input sound pressure level at the microphone; measuring for a given measurement parameter set of perceived loudness levels and frequencies or frequency bands the respective transducer input audio signal level to be applied to the transducer input in order to achieve the respective perceived loudness level at the respective frequency or frequency band, said measurement parameter set comprising at least a low loudness level, an intermediate loudness level and a high loudness level, and said intermediate loudness level being measured for a larger number of frequencies or frequency bands and with a finer frequency resolution than said low and high loudness levels; calculating an individual gain function to be implemented in the audio signal processing unit in order to achieve the pre-defined target loudness function by taking into account the measured transducer input audio signal levels; and operating the hearing instrument with the individual gain function.