377 results about "Frequency Unit" patented technology

A method for the partitioning frequency resources used in the transmission of control signals and data signals by user equipments in a communication system. The control signals and data signals are for periodic transmission and dynamic transmission. Also provided is an apparatus and method for user equipments to determine the first frequency unit available for the transmission of dynamic control signals, such as acknowledgement signals associated respective reception of data signals configured through a scheduling assignment by a serving Node B. The utilization of the operating bandwidth is maximized by avoiding fragmentation and facilitates the achievement of reception reliability targets particularly for control signals.

The present specification relates to an apparatus and method for transceiving signals between a terminal and a base station in a wireless communication system. The present specification relates to a signal-transceiving method in which location reference signals discriminated by frequency units for each base station, such that base stations which transmit location reference signals with the same location reference signal pattern can be further discriminated.

An system and method for transmitting/receiving sounding resource allocation information in a communication system are provided, in which a transmitter selects one of a plurality of preset sounding resource allocation patterns according to operation mode information about an receiver to which uplink resources will be allocated, the uplink resources having a period of a predetermined number of time units and a predetermined number of frequency units, reads resource layout information about each sounding subblocks of an active sounding resource allocation pattern corresponding to the selected sounding resource allocation pattern, and allocates at least one of the sounding subblocks of the sounding resource allocation pattern to the receiver according to the resource layout information and transmitting a sounding resource allocation information including the allocated sounding subblock information.

The invention discloses a rolling bearing fault feature extraction method based on the signal sparse representation theory, and the method comprises the following steps: constructing an over-complete dictionary representing local damages of a rolling bearing through employing a multi-stage inherent frequency unit impulse response function; recognizing the multi-stage inherent frequency and damping ratio of the rolling bearing and a sensor system from a vibration response signal through a related filtering method, and obtaining an optimized dictionary; solving a sparse coefficient through employing a matching tracking algorithm, and improving the solving speed and precision through reasonable segmentation; reconstructing an impact response signal of each segment, and obtaining the sparse representation of a fault feature signal; carrying out time domain index statistic characteristic analysis of time intervals of adjacent impact response components in a sparse signal, and diagnosing the type of a fault through combining a mean value and a mean square deviation value. The method has the advantages of an analytical method and an adaptive method, improves the precision of waveform features, and can iron out the defects that a conventional method based on Fourier transform is not suitable for rotating speed fluctuation.

The invention discloses a voice enhancement method for fusing phase estimation and human ear hearing characteristics in a digital hearing aid, comprising steps of obtaining a frequency domain expression mode containing noise through Fourier transformation, adopting a minimum value control recursive average method to obtain a noise power spectrum, obtaining initial enhancement voice and a noise amplitude spectrum, obtaining the initial enhancement voice and noise through correcting the phase of the voice and the noise through improving the phase estimation of the voice distortion under the low signal-to-noise ratio environment, performing filtering processing on the initial enhancement voice and the noise through a gammatone filter assembly which simulates the working mechanism of the artificial cochlea, performing analysis on the time frequency of the gammatone filter assembly to obtain the time frequency expression form consisting of time frequency units, using the hearing characteristics of the human ear to calculate binary mast containing noise in the time frequency domain, and using the mask value to obtain the enhanced voice after synthesis.

The invention relates to the field of voice technologies, in particular to a self-adaptive denoising method based on sub-band noise analysis. The method includes the steps that firstly, framing and short time frequency domain transformation are conducted on input time domain audio signals with noise, and then frequency domain audio signals with noise are generated; secondly, a noise energy spectrum of the frequency domain audio signals with noise is estimated through a minimum value tracking method; thirdly, the posterior signal to noise ratio and the prior signal to noise ratio of the noise energy spectrum are calculated; fourthly, through a nonlinear gain extension method, denoising gains of all time frequency units are calculated through the posterior signal to noise ratio and the prior signal to noise ratio; fifthly, smoothing filtering is conducted on the denoising gains of all the time frequency units to reduce tone quality distortion; sixthly, the denoising gains act on all the time frequency units of the audio signals with noise in the first step, and then denoised frequency domain audio signals are acquired; seventhly, short time frequency domain inverse transformation is conducted, and then the final denoised time frequency audio signals are acquired and output. According to the method and system, stable noise in target signals can be greatly lowered.

The invention discloses a method and a system for capturing a weak GNSS (Global Navigation Satellite System) signal under the condition of large-scale frequency deviation. The method comprises the following steps of: according to a pre-detection integration time, a carrier frequency and a sampling rate, determining a maximum carrier frequency searching step; dividing the whole frequency range to be searched into a plurality of carrier frequency units according to the maximum frequency searching step to realize rough search of the carrier frequency; mapping a received satellite signal onto each carrier frequency unit and regulating a local pseudo code rate according to the carrier frequency units so as to reduce the influence of code Doppler; calculating correlation in a flexible zero-padding mode through FFT (Fast Fourier Transform)/IFFT (Inverse Fast Fourier Transform) so as to realize code phase parallel search; and carrying out FFT operation on different correlation values corresponding to the same code phase so as to fulfill the aim of carrying out fine search on the carrier frequencies in the carrier frequency units. By the method and the system, a capturing speed and a capturing sensitivity of a receiver can be improved, so that the receiver completes the work of capturing a weak navigation signal pseudo code under the condition of the large-scale frequency deviation.

The invention discloses a multi-speaker voice separation method based on a convolutional neural network and depth clustering. The method comprises the following steps: 1, a training stage: respectively performing framing, windowing and short-time Fourier transform on single-channel multi-speaker mixed voice and corresponding single-speaker voice; and training mixed voice amplitude frequency spectrum and single-speaker voice amplitude frequency spectrum as an input of a neural network model; 2, a testing stage: taking the mixed voice amplitude frequency spectrum as an input of a threshold expansion convolutional depth clustering model to obtain a high-dimensional embedded vector of each time-frequency unit in the mixed frequency spectrum; using a K-means clustering algorithm to classify thevectors according to a preset number of speakers, obtaining a time-frequency masking matrix of each sound source by means of the time-frequency unit corresponding to each vector, and multiplying thematrixes with the mixed voice amplitude frequency spectrum respectively to obtain a speaker frequency spectrum; and combining a mixed voice phase frequency spectrum according to the speaker frequencyspectrum, and obtaining a plurality of separate voice time domain waveform signals by adopting short-time Fourier inverse transform.

The invention provides a single-channel voice enhancement method and a single-channel voice enhancement system. The method comprises the following steps of: extracting a noise signal from a noisy voice signal through voice activity detection; respectively performing outer ear, inner ear and middle ear simulation manipulation to the noisy voice signal and the noise signal through peripheral analysis; obtaining energy difference of each time frequency unit of the noisy voice signal and the noise signal subjected from simulation manipulation through feature extraction; generating different masking values to the energy difference of each time frequency unit and weighing the different masking values to obtain a masking processing signal; and rebuilding the voice signal to the masking processing signal and the noisy voice signal subjected from simulation manipulation to obtain a voice enhancement signal. The invention can decrease damage to a target voice signal and realize better denoising effect and keep higher voice quality under the environment with multi noises.

The invention discloses an uplink signal sending method and device. The uplink signal sending method includes generating to-be-sent uplink demodulation reference signals in a first time frequency unit, wherein the frequency domain of the first time frequency unit is a single sub carrier or a plurality of sub carriers; mapping the uplink demodulation reference signals to a first time frequency position of the first time frequency unit; sending the uplink demodulation reference signals in the first time frequency unit and uplink data corresponding to the uplink demodulation reference signals. According to the invention, through reasonable design of the uplink data sending method, a problem that the uplink demodulation reference signals cannot be sent by taking the sub carrier as the modulation unit in the prior art is solved.

The invention discloses a Ku frequency band double-frequency dual-polarization micro-strip plane reflective array antenna. A plane medium substrate is provided with two kinds of frequency passive micro-strip resonance units, each kind of frequency units is arranged in equal rows and equal lines, the distance between every two rows is equal to that between every two lines, the distance is less than or equal to one half of the wavelength of the highest work frequency free space, and each second frequency unit is located at the center of every four first frequency units in a square array. The center of all the first frequency units is provided with a set of gaps, the center of all the second frequency units is provided with a set of gaps perpendicular to the gaps of the first frequency units, and polarization in the two perpendicular directions is formed. Each kind of frequency units is of loop line patch structures with at least two layers, and the size of a loop is determined according to the phase position needing compensation. According to the Ku frequency band double-frequency dual-polarization micro-strip plane reflective array antenna, the phase difference between a feed source and the space of each unit on the array surface is compensated by regulating the size of each micro-strip unit on the medium substrate, reflected waves achieve same-phase superposition in the special direction, pencil beams are formed, and the purpose of high gain is achieved.

The invention discloses a sparse-decomposition-based hybrid fault feature extraction method of a gear wheel and a bearing, wherein the method can be used for diagnosing a hybrid fault formed by a distributed gear wheel fault and a local gear wheel and bearing fault in a gear case. When a steady modulation dictionary is constructed, atomic parameter optimization is carried out by using a discrete frequency spectrum correction technology, thereby improving precision of steady modulation component separation. When an impact modulation dictionary is constructed, an over-complete dictionary using a multi-stage inherent-frequency unit impulse response function as an atom is established and the inherent frequency and the damping ratio are identified in a self-adapting mode from a fault vibration signal, so that an impact response waveform caused by local faults of the gear wheel and the bearing can be represented well. After optimization of the steady modulation dictionary and the impact modulation dictionary, the dictionary redundancy is substantially reduce; and with a segmented matching tracking method, the point number of inner product calculation during the sparse coefficient solving process is reduced. On the basis of the two kinds of measures, the speed of signal sparse decomposition is improved.

The invention provides a speech separation method based on a fuzzy membership function, and belongs to speech separation methods. The fuzzy membership function is combined in the speech separation method, so that more accurate definition of a membership degree of speech time frequency units to a target signal is obtained. An auditory oscillation model is built through human ear auditory system simulation, and speech pitch characteristics are extracted. The speech time frequency units are marked according to pitch cycle characteristics to form foreground streams and background streams. Whether the corresponding time frequency units are targets or noise is judged according to different marks. In the synthesis stage, a target unit multiplies a high weight, a noise unit multiplies a low weight, and resynthetized speech is obtained. By means of the speech separation method, the pitch cycle can be estimated more precisely, the time frequency units can be marked more accurately on the basis of characteristic clues, and the more complete target speech can be obtained. Due to the fact that the method is based on the pitch characteristics of the speech, good separation effects in complex and non-stationary noise are achieved, and the application range is wide.

First, the method ensures pairwise grouping scheme of multiple sub band resources belonged to same user on same antenna. Next, the method carries out orthogonal transform (OT) for data symbols on same time frequency unit in two sub bands of each group, and then sends the transform. The invention also discloses device for realizing sending data through multiple channels in multicarrier system. The device includes following units: sub band map unit in use for mapping data symbols belonged to same user to multiple sub bands assigned for the user, and informing the utilized information of sub band to the grouping unit; the grouping unit in use for carrying out pairwise grouping multiple sub band resources belonged to same user on same antenna, and informing grouping scheme to OT unit; the OT unit is connected the map unit, and its function is described above.

The present invention discloses a method for measuring a carrier interference noise ratio. The method comprises that a plurality of sub-carriers are respectively selected from the frequency spot on the two symbols in one or a plurality of time frequency units, and the number of the sub-carriers selected from each time frequency unit is the same; one or two symbols is (are) respectively selected from the two symbols of the time frequency unit, the interference noise power of the frequency domain channel response estimated value of the sub-carrier on the symbols selected is accounted, and the interference noise power of the sum of the frequency domain channel response estimated value of the sub-carriers on the symbol; according to the characteristic that the frequency domain channel response of the sub-carrier on the symbol in the time frequency unit keeps invariable on the frequency domain, the power error caused by the frequency domain channel response changing along time in the former interference noise power is eliminated; according to the interference noise power of the sub-carrier frequency domain channel response estimated value after being eliminated, the carrier interference noise ratio is calculated. The present invention can accurately measure the carrier interference noise ratio when the terminal experiences a time-varying channel or a time-invariant channel.

The invention discloses a narrow-band interference detecting method and device in an MIMO-OFDM (Multiple Input Multiple Output-Orthogonal Frequency Division Multiplexing) system. The method comprises the follow steps of: a pilot-frequency sub-carrier group composed of two pilot frequencies is selected in each time-frequency unit, and the two pilot frequencies in the pilot-frequency sub-carrier group are on different frequency positions of different OFDM signs; and the channel response values of two sub-carriers in the pilot-frequency sub-carrier group minus the signal power sum of the pilot-frequency sub-carrier group obtain the interference noise power value of the pilot-frequency sub-carrier group, so that the interference noise power value of each sub-carrier of the time-frequency unit is determined. The device comprises a pilot-frequency sub-carrier group selecting module and an interference noise power value determining module. The method and the device can accurately estimate the narrow-band interference information and can accurately detect the narrow-band interference position and power even if the narrow-band interference is weaker.

In a terminal (200), a transmission control unit (206) identifies candidate resources on the basis of setting information and determines, on the basis of a trigger signal, a mapping resource, in which a second type of reference signal is mapped, in candidate frequency unit groups constituting the candidate resources. Each of the candidate resources is constituted by a plurality of fundamental subframe groups that are arranged at first intervals. N constituent subframes constituting each of the fundamental subframe groups are arranged at second intervals narrower than the first intervals. Each of the candidate frequency units of each candidate frequency unit group is a frequency unit specified by a predetermined frequency hopping pattern in M frequency units in each constituent subframe.

Wind turbine comprising a nacelle with a generator and wind turbine blades for driving the generator, a tower supporting the nacelle wherein the nacelle is rotatable about a vertical axis an azimuth drive for rotating the nacelle around the axis, said azimuth drive comprising a plurality of asynchronous yaw motors and an azimuth controller with a frequency unit for controlling the plurality of asynchronous yaw motors, wherein the azimuth controller is arranged to connect all of the plurality of asynchronous yaw motors to the one frequency unit in a rotating mode and to connect only a subset of the plurality of asynchronous yaw motors from the one frequency unit in a locking mode.

The invention provides an LINC transmitter, belongs to the technical field of wireless communication, and relates to a transmitting apparatus in wireless communication technology, in particular to a linear amplification with nonlinear components (LINC)-based transmitter. The LINC transmitter comprises a signal separating unit, a variable-frequency unit, a signal amplifying unit, a signal synthesizing unit, a radio-frequency antenna and a feedback control unit, wherein the feedback control unit couples partial output signals; the partial output signals are subjected to attenuation and down-conversion and then compared with an input signal to obtain an error signal; the error signal is utilized to perform real-time regulation and control on the output power of any power amplifier in the signal amplifying unit so as to realize gain balance between a first switch type power amplifier and a second switch type power amplifier in the signal amplifying unit. The output power of the switch type power amplifiers in the LINC transmitter is fed back and controlled and the gain imbalance between two paths of signals are eliminated so as to improve the linearity of the LINC transmitter and enable the LINC transmitter to realize high-efficiency and high-linearity amplification on a wireless communication signal.

The invention provides a detection method and equipment for an underwater target. The detection method includes the steps: receiving received signal of each array element in a towed line array, conducting fast Fourier transform analysis to received signal of each array element, and acquiring spatial spectrum of each frequency unit, taking a maximum amplitude value of spatial spectrum of the each frequency unit; by means of the maximum amplitude value, conducting amplitude value weighting to spatial spectrum of the each frequency unit; and finally conducting accumulation for a sum to the spatial spectrum of the each frequency unit post-weighting. When the target radiating line spectrum signal is weak, compared with the untreated by amplitude value weighting, the detection property of the amplitude-weighting-based target detection method can be effectively improved.

A vice enhancement method based on double-ear sound source positioning and deep learning in a double-ear digital hearing aid belongs to the field of voice signal processing. Firstly a two-stage deep neural network is used for accurately positioning a target voice, and noise in a direction which is different from the direction of target voice is eliminated according to spatial filtering. By means of a deep learning model in which a time delay control bidirectional long-short term memory deep neural network and a classifier are combined, an extracted multi-resolution hearing cepstrum coefficientis used as a characteristic input. Through nonlinear processing capability of deep learning, each time frequency unit of the noise-containing voice is classified to a voice time frequency unit or noise time frequency unit. Finally a voice waveform combining algorithm is used for eliminating the noise in the direction which is same with that of the target voice. The algorithm eliminates the noisein the direction which is different from the direction of the target voice and eliminates the noise in the direction that is same with the target voice, and finally obtains the enhanced voice which satisfies speech intelligibility and comfort of a deaf person. All deep learning models utilize offline training, thereby satisfying a requirement for real-time performance.