323results about How to "High frequency resolution" patented technology

A digitizer for user interaction via an object with an electronically refreshable display screen, the digitizer comprising: a transparent sensing arrangement of detectors located at said electronically refreshable display screen for detecting an electric field of said object, said detectors having outputs, and an arrangement of differential amplifiers associated with said outputs, thereby to apply differential detection between said outputs.

An improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.

The present invention provides a new method and an apparatus for spectral envelope encoding. The invention teaches how to perform and signal compactly a time / frequency mapping of the envelope representation, and further, encode the spectral envelope data efficiently using adaptive time / frequency directional coding. The method is applicable to both natural audio coding and speech coding systems and is especially suited for coders using SBR [WO 98 / 57436] or other high frequency reconstruction methods.

A novel bandwidth extension technique allows information to be encoded and decoded using a fractal self similarity model or an accurate spectral replacement model, or both. Also a multi-band temporal amplitude coding technique, useful as an enhancement to any coding / decoding technique, helps with accurate reconstruction of the temporal envelope and employs a utility filterbank. A perceptual coder using a comodulation masking release model, operating typically with more conventional perceptual coders, makes the perceptual model more accurate and hence increases the efficiency of the overall perceptual coder.

A digitizer for user interaction via an object with an electronically refreshable display screen, the digitizer comprising: a transparent sensing arrangement of detectors located at said electronically refreshable display screen for detecting an electric field of said object, said detectors having outputs, and an arrangement of differential amplifiers associated with said outputs, thereby to apply differential detection between said outputs.

This invention provides a method and a system for coding / decoding frequency domain parameter stereos and multiple tracks, which converts signals of right and left tracks to a residual signal of a sum-difference tack and a residual signal of an estimated difference track, carries out time frequency conversion to residual signals of the estimated difference track, the original difference track and the original sum track to pick up the HF component and low frequency component of the residual signal of the original difference track, which is coded in quantization to utilize the HF component to get estimated residual RF component of the right and left tracks and gets the original residual RF component of the right and left tracks to be controlled in gain to get a gain coefficient of the difference of the two tracks and coded in quantization and finally multiplexes the coded data and edge information to get a frequency domain parameter stereo bit stream.

A post-processing method for a speech decoder which outputs a decoded speech signal in the time domain provides high frequency resolution based on a frequency spectrum having non-harmonic and noise deficiencies. This is obtained by transforming the decoded time domain signal to a frequency domain signal by using a high frequency resolution transform (FFT). Then an analysis of the energy distribution of the frequency domain signal is made throughout its frequency area (4 kHz) to find the disturbing frequency components and to prioritize such frequency components which are situated in the higher part of the frequency spectrum. Next, the suppression degree for the disturbing frequency components is found based on prioritizing. Finally the steps of controlling a post-filtering of the transform in dependence of the finding, and inverse transforming the post-filtered transform in order to obtain a post-filtered decoded speech signal in the time domain are performed.

There is provided a tempo detection apparatus capable of detecting, from the acoustic signal of a human performance of a musical piece having a fluctuating tempo, the average tempo of the entire piece of music and the correct beat positions, and further, the meter of the musical piece and the position of the first beat. The tempo detection apparatus includes an input section 1 for receiving an acoustic signal; a chromatic-note-level detection section for applying an FFT calculation to the received acoustic signal at predetermined time intervals to obtain the level of each chromatic note at each of predetermined timings; a beat detection section 2 for summing up incremental values of respective levels of all the chromatic notes at each of the predetermined timings, to obtain the total of the incremental values of the levels, indicating the degree of change of entire sound at each of the predetermined timings, and for detecting an average beat interval and the position of each beat from the total of the incremental values of the levels; and a measure detection section 3 for calculating the average level of each chromatic note for each beat, for summing up incremental values of respective average levels of all the chromatic note for each beat to obtain a value indicating the degree of change of entire sound at each beat, and for detecting a meter and the position of a measure line from the value indicating the degree of change of entire sound at each beat.

The invention relates to a system and a method for measuring time-sharing quick steady-state sine sweep excitation frequency response function, which belong to the technical field of vibration tests. An excitation system adopts a vibration exciter and a power amplifier. A force sensor and a vibration sensor are adopted to measure an exciting force signal and a vibratory response signal of the system at the same time. A charge amplifier is adopted to condition and filter the signals. A data acquisition system connects a data acquisition card with a connecting terminal through a high-performance shielded cable and is connected with a general computer to acquire and send data. The generation of a sweep signal and the data processing, analysis, storage and display of an exciter response signal are achieved by adopting software based on the general computer. The system and the method achieve quick steady-state sine sweep excitation, the complete cycle sampling of the exciter response signal, and precise and reliable frequency response function curve. The test system is guided by computer programs and is finished automatically, the test method is precise and reliable, and the system and the method have important meaning on testing the dynamic properties of structural elements.

The invention discloses an automobile forward direction anticollision system signal processing method. In each frequency sweep cycle, the following processes which are performed on upper sweep frequency IQ data and lower sweep frequency IQ data of each channel comprise steps of performing fast Fourier transform FFT on N points of IQ data, performing adaptive threshold detection on transformed plural module value data after transmission, outputting an over-threshold point position, taking an interval between two points as a transformation interval, wherein one point is positioned before the threshold point position and the other one is positioned after the threshold point position, performing chirp-z transformation on M points, performing peak value searching on the data after chirp-z transformation, outputting a peak value point position, calculating a difference frequency and a phase according to the peak value point, outputting an object distance, relative radial speed and an azimuth according to the difference frequency and the phase output distance. The invention enables the system to respond fast through processing and optimizing the signal, is good in practicability and faster.

a Method and Apparatus for Automatic Jammer Frequency Control of Surgical Reactive Jammers is disclosed. The system and method to autonomously and selectively jams frequency-hopping signals in near real-time. The system has the ability to automatically detect short duration signals (such as those output from frequency hoppers), to automatically determine if detected signal(s) should be jammed, and subsequently to automatically and extremely quickly activate the jamming transmitter on the frequency-hopper transmitter's frequency. Furthermore, the system improves upon the FFT-based front-end system by incorporating a module that automatically and iteratively comparing the digital FFT results to the analog RF signal to arrive at a total frequency offset between the FFT results and the analog signal so that the transmitted jamming signal will be more accurately applied on the desired or target frequency. Finally, the system provides a programmable user interface so that operators can set up the system to act autonomously as intended, such that operator intervention is unnecessary when the system is placed in jamming operation mode.

Acoustic signal encoder is provided which comprises a subband filter band to divide an original signal into a plurality of frequency bands, a spectrum transformation circuit to detect the amplitude of a signal in each of the plurality of frequency bands in each of sub-blocks resulted by division of a block length for signal coding, process the signal amplitude in each band based on the detected amplitude and transform the signals divided in the frequency bans to spectra, a normalizing circuit and quantizing circuit to normalize and quantize the spectrum, respectively, and a code row generator to generate a code row from the signals processed by the above circuits.

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.

The invention relates to encoding of broadband and narrowband acoustic source signals (x) such that the perceived sound quality of corresponding reconstructed signals is improved in comparison to the known solutions. An enhancement estimation unit (102), operating in serial or in parallel with the regular encoding / decoding means (101), perceptually enhances a reconstructed acoustic source signal by utilization of an enhancement spectrum (C) comprising a larger number of spectral coefficients than the number of sample values in corresponding frames of the signals carrying the basic encoded representation of the acoustic source signal. The thus extended block length of the enhancement spectrum frame provides a basis for accomplishing the desired improvement of the perceived sound quality.

The invention provides a frequency calibration circuit and a method thereof. The frequency calibration circuit comprises a phase lock loop circuit, an M-dividing divider for generating a plurality of phase signals, a sampling circuit for sampling on the rising edges of the phase signals, a first counter for counting reference frequency signals, a second counter counting first phase signals, a logical control circuit for comparing a count value with a target count value to obtain a minimum difference value, and looking up a control bit of a voltage-controlled oscillator through the difference value, and a comparison circuit for judging whether the control voltage of the voltage-controlled oscillator is in a set range or not. The method comprises the following steps: sampling the reference frequency signals to obtain a total count value; comparing the total count value with a target count value; selecting a next resonance line according to a binary lookup flow till binary lookup ends; and outputting a resonance line corresponding to a minimum error in order that the output frequency of the voltage-controlled oscillator reaches a required value. Through adoption of the frequency calibration circuit and the method thereof, the speed and accuracy are increased, and the requirement on the working frequency of a digital circuit is lowered, so that a design method is simplified.

The invention discloses a power system interharmonic wave detection method based on MUSIC spectrum estimation and a HBF neural network, comprising the steps of getting sampling data of detected signals by an analog-to-digital converter, getting various harmonic waves and interharmonic wave frequency contained in the signals through MUSIC spectrum estimation, and getting amplitudes and phases of the harmonic waves and the interharmonic waves through the adaptive adjustment of weights of the HBF neural network. The invention first provides the power system interharmonic wave measurement method based on the MUSIC spectrum estimation and the HBF neural network. The invention has the advantages of high frequency resolution without causing frequency spectrum leakage, high measurement precision of frequency, amplitudes and phases of the harmonic waves and the interharmonic waves, and high learning speed of the weights of the neural network, and is adaptive to on-line monitoring and off-line testing of interharmonic waves in the power system.

The present invention provides a new method and an apparatus for spectral envelope encoding. The invention teaches how to perform and signal compactly a time / frequency mapping of the envelope representation, and further, encode the spectral envelope data efficiently using adaptive time / frequency directional coding. The method is applicable to both natural audio coding and speech coding systems and is especially suited for coders using SBR [WO 98 / 57436] or other high frequency reconstruction methods.

The invention relates to a multi-robot system fault diagnosis method, comprising the following steps of: 1) obtaining real-time robot motion data; 2) adopting wavelet packet transform to perform feature extraction on the motion data; 3) inputting to-be-diagnosed data subjected to feature extraction into a trained fault diagnosis model, and calculating a corresponding real-time likelihood probability of current robot motion data; and 4) according to a relation between the real-time likelihood probability and a state threshold value, obtaining a current hidden state of the multi-robot system and obtaining a fault diagnosis result. Compared with the prior art, the method has the advantages of high robustness, accurate diagnosis result, wide application range and the like.

A piezoelectric ceramic dual-band bass-enhanced earpiece includes an earphone shell, a moving coil speaker assembly and a piezoelectric ceramic speaker component. The earphone shell includes an acoustic output orifice. The moving coil speaker assembly includes a moving coil speaker and a circuit connecting unit electrically connected to the moving coil speaker. The piezoelectric ceramic speaker component is located between the acoustic output orifice and moving coil speaker assembly and includes a piezoelectric ceramic plate and a conductive member connecting to the piezoelectric ceramic plate and the circuit connecting unit. The piezoelectric ceramic plate includes a plurality of acoustic through orifices. With the combination of the piezoelectric ceramic speaker component and moving coil speaker assembly, the piezoelectric ceramic dual-band bass-enhanced earpiece achieves the hearing effect that high-frequency sound wave surrounded by low-frequency sound wave. Due to increasing high-frequency resolution, the user can feel the improvement of bass frequency effect.

In a method for suppressing a noise by the spectrum subtraction method, it is possible to improve the noise suppression capability by simultaneously obtaining a frequency resolution required for the noise estimation spectrum and a temporal resolution required for the noise suppression spectrum. The signal length of an observation signal cut out for analyzing the spectrum of the observation signal used for estimation calculation of the noise spectrum is set longer than the signal length of an observation signal cut out for analyzing the spectrum of the observation signal as a value to be subtracted for performing subtraction with the noise spectrum.

The present document relates to methods and systems for music information retrieval (MIR). In particular, the present document relates to methods and systems for extracting a chroma vector from an audio signal. A method (900) for determining a chroma vector (100) for a block of samples of an audio signal (301) is described. The method (900) comprises receiving (901) a corresponding block of frequency coefficients derived from the block of samples of the audio signal (301) from a core encoder (412) of a spectral band replication based audio encoder (410) adapted to generate an encoded bitstream (305) of the audio signal (301) from the block of frequency coefficients; and determining (904) the chroma vector (100) for the block of samples of the audio signal (301) based on the received block of frequency coefficients.

The invention provides a cochlea stimulator for implantation comprising optical fibres of which are coupled to an irradiation source that is controlled by a modulator to generate irradiation specific for a pre-determined range of sound-frequencies. The cochlea stimulator effects a frequency-specific activation of the organ of Corti needed for speech perception especially in noisy environment and more complex sounds. For imparting excitation signals which are generated by modulated pulsed laser irradiation conducted within an optical fibre in order to elicit nervous signals in residual functional organ of Corti sections, the auditory prosthesis preferably contains optical fibres which are dimensioned to terminate in end sections within the cochlea at different sites or sections of the organ of Corti. e.g. having different lengths for locating their end sections at different internal parts of the cochlea.

The invention provides a modularized frequency synthesizing method based on direct digital and direct analogue combination, including the following steps: (1) direct digital frequency synthesizing method is adopted to generate M intermediate frequency marker signals with frequency interval of deltaF and bandwidth of (M-1)deltaF, namely F1, F1+deltaF, F1+2deltaF, ..., F1+(M-1)deltaF; (2) frequency shifting method is adopted to shift the frequencies of the M intermediate frequency marker signals generated in the step (1) to microwave frequency band, so as to generate M microwave local oscillation signals with frequency interval of deltaF and bandwidth of (M-1)deltaF, namely F2, F2+deltaF, F2+2deltaF, ..., F2+(M-1)deltaF; (3) indirect analogue frequency synthesizing method is adopted to generate N radio frequency marker signals with frequency interval of MdeltaF and bandwidth of (N-1)MdeltaF, namely F3, F3+MdeltaF, F3+2MdeltaF, ..., F3+(N-1)MdeltaF; (4) M microwave local oscillation signals generated in step (2) and N radio frequency marker signals generated in step (3) are subject to mixing, so as to obtain M*N frequency synthesis signals with frequency interval of deltaF and bandwidth of (MN-1)deltaF, namely F2+F3, F2+F3+deltaF, F2+F3+2deltaF, ..., F2+F3+(MN-1)deltaF, and the frequency synthesis signals are output. The invention realizes extremely rapid frequency conversion, extremely low phase noise and extremely low spurious level with low cost.

A signal processing apparatus includes a signal output unit for outputting a measurement signal, the measurement signal being produced by synthesizing a signal composed of a concatenation of 2d period signals with a sinusoidal signal, each period signal having a time-domain waveform period being 2n samples, the sinusoidal wave having a wave count within the concatenation period of 2d period signals being other than an integer multiple of 2d, and n and d being respectively natural numbers, and an analyzing unit for frequency analyzing a response signal obtained as a result of picking up the measurement signal output from the signal output unit.

A decoder for generating an un-mixed audio signal including a plurality of un-mixed audio channels is provided. Moreover, an encoder and an encoded audio signal is provided. The decoder includes an un-mixing-information determiner for determining un-mixing information by receiving first parametric side information and second parametric side information on the at least one audio object signal, wherein the frequency resolution of the second parametric side information is higher than that of the first parametric side information. Moreover, the decoder includes an un-mix module for applying the un-mixing information on a downmix signal, to obtain an un-mixed audio signal including the plurality of un-mixed audio channels. The un-mixing-information determiner is configured to determine the un-mixing information by modifying the first parametric information and the second parametric information, such that the modified parametric information has a frequency resolution which is higher than the first frequency resolution.

The invention discloses a stable underwater communication node awakening signal detection method. The stable underwater communication node awakening signal detection method comprises the following steps of: S1, emitting a dual-frequency or multi-frequency awakening signal; S2, according to a set initial frequency, transform point number and frequency resolution ratio, executing CZT (chirp z transform) on a unit circle on the awakening signal; S3, searching an extreme value of a zoom spectrum of CZT, judging a signal as the awakening signal if an estimated relative frequency interval and relative amplitude relationship between frequency components are kept unchanged, or judging the signal is not the awakening signal; and S4, after the signal is determined as the awakening signal, powering on an integral communication node to enable the node to be transferred into a working state, and when the node completes working, cutting off a power supply to enable the node to be transferred into a sleeping state. According to the stable underwater communication node awakening signal detection method disclosed by the invention, due to the high frequency resolution ratio, multiple groups of frequencies can be adopted for combination to awaken a plurality of nodes of an underwater sensor network without worrying about mutual interference, so that detection false-alarm is greatly reduced and a service life of the underwater wireless sensor network is prolonged.

A magnetic device includes: a magnetoresistive effect element having a magnetization fixed layer, a magnetization free layer, and a nonmagnetic layer sandwiched between the magnetization fixed layer and the magnetization free layer; an input terminal for feeding an AC signal to the magnetoresistive effect element in its stacking direction; and an output terminal for extracting an output voltage from the magnetoresistive effect element, wherein the nomagnetic layer includes an insulating layer portion comprising an insulating material, and a current-constricting layer portion comprising a conductive material which passes through the insulating layer portion in its film thickness direction.