CN112804180A - Amplitude limiting OQAM/FBMC system signal transceiving method based on compressed sensing - Google Patents
Amplitude limiting OQAM/FBMC system signal transceiving method based on compressed sensing Download PDFInfo
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- H04L27/00—Modulated-carrier systems
- H04L27/26—Systems using multi-frequency codes
- H04L27/2601—Multicarrier modulation systems
- H04L27/2697—Multicarrier modulation systems in combination with other modulation techniques
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Abstract
The invention belongs to the technical field of information and communication, and particularly relates to an OQAM/FBMC system signal receiving and transmitting method based on amplitude limiting of compressed sensing. The invention has sparsity in time domain according to the distortion caused by amplitude limiting processing on OQAM/FBMC signals, namely, only a small part of the signal distortion in the time domain is a non-zero value, and the distortion can be well estimated and compensated by utilizing a Compressive Sensing (CS) method, thereby greatly reducing the PAPR under the condition of ensuring the error rate of a system. Estimating a truncated portion z of the signal while effectively reducing the PAPR of the signal to more accurately recover the signal; the invention carries out the Clipping and Filtering processing and uses the Turbo-CS algorithm to estimate the distorted signal in the time domain, on one hand, the PAPR of the signal is reduced by the Clipping at the transmitting end and the Filtering is used to relieve the side lobe regeneration, on the other hand, the Turbo-CS algorithm is used at the receiving end to detect the distorted signal z to ensure the performance of the error rate of the system.
Description
Technical Field
The invention belongs to the technical field of information and communication, and particularly relates to an OQAM/FBMC system signal receiving and transmitting method based on amplitude limiting of compressed sensing.
Background
As an alternative technology to OFDM, an Offset Quadrature Amplitude Modulation based Filter Bank Multi-carrier (OQAM/FBMC) has the advantages of low out-of-band leakage of frequency spectrum, low sensitivity to synchronization deviation, and higher robustness to multipath channels, and is currently widely researched. However, like other multi-carrier modulation techniques, the OQAM/FBMC technique also has a high PAPR problem, and the PAPR problem becomes more complicated due to the overlapping of the OQAM/FBMC symbols. At present, means for reducing PAPR of an OQAM/FBMC signal mainly include a signal distortion and distortion-free reduction method, the distortion-free method needs to overcome the problem of obtaining an optimal reduction effect under a low complexity condition and simultaneously reduce overhead, and the signal distortion reduction method needs to avoid a side lobe regeneration phenomenon caused by signal distortion and reduce in-band distortion of the signal as much as possible.
A simple and effective method is to clip the signal in advance, i.e. before the carrier signal passes through the high power amplifier, etc., the time domain signal is clipped, if the peak value of the signal exceeds a given threshold value, the signal is set to a preset threshold value while the phase is unchanged, and if the peak value of the signal is not greater than the threshold value, no operation is performed.
Where a is a set amplitude threshold value,representing the phase of the symbol. The clipping method is simple to operate and low in computational complexity, and can significantly improve the PAPR performance of the system, but belongs to distortion transformation, and can cause serious nonlinear distortion to signals. .
Disclosure of Invention
In order to solve the problems in the prior art, the distortion caused by amplitude limiting processing on the OQAM/FBMC signal has sparsity in the time domain, namely only a small part of the signal distortion in the time domain is a non-zero value, and the distortion can be well estimated and compensated by using a Compressive Sensing (CS) method, so that the PAPR is greatly reduced under the condition of ensuring the system error rate. Therefore, the signal transceiving method of the OQAM/FBMC system based on amplitude limiting of compressed sensing is provided.
The receiver of the invention consists of three modules, namely a linear estimation module A, a signal demodulation module B and an amplitude limiting estimation noise reduction module C. Each module outputs an estimation and transmits the estimation to the next module, the OQAM/FBMC symbol a is iteratively estimated by the module A and the module B, the sparse distortion z is iteratively estimated by the module A and the module C, and iteration is carried out among the modules until the algorithm is converged.
The technical scheme adopted by the invention is as follows: a signal transceiving method of an OQAM/FBMC system based on compressed sensing amplitude limiting comprises the following steps:
s1, inputting binary bit stream b [ k ]]QAM modulating to obtain a mapping symbol c with length Mm,nAs an OQAM/FBMC symbol, two real symbols with the length of M are obtained by taking a real part and an imaginary partAndnoting that the nth transmitted symbol isCarrying out phase rotation, frequency domain filtering and inverse fast Fourier transform on the transmitted symbol to obtain a length LgUp-sampled signal vector s of KM (K is an overlap factor indicating the number of overlap of OQAM/FBMC symbols)n,
sn=FHGΘnan
Wherein the filter matrix isGmIs the length L on the m sub-carriergFrequency domain response, phase rotation of prototype filterθm,n=ejπ(m+n)/2ejnmπF is Lg×LgUnit DFT matrix, (.)HRepresenting a conjugate transpose.
S2, taking N OQAM/FBMC symbols as a data frame, delaying the generated 2N signal vectors by half a symbol period, namely M/2 sampling points in turn, generating a transmitting signal after superposition,
s=Wa
whereinLet Wn=FHGΘnThen the modulation matrix isThe internal structure is shown in figure 1, and the total length of the transmitted signal isThe s is clipped and filtered and then transmitted to the transmitting antenna.
And S3, the signal is transmitted by the transmitting antenna, the signal passes through a multipath channel, and the tail value of the prototype filter is close to 0, so the process is a process of circularly convolving the signal and the channel.
S4, the signal arrives at the receiving end to obtain the length LsThe observed signal of (a) is,
r=HWa+Hz+n
wherein H is a channel response matrix, z is an interference signal generated by the Clipping and filtering operation, n is zero-mean Gaussian white noise, and the noise variance is
S5, the receiving end cancels the interference and estimates the symbol a, specifically: the receiver comprises a linear estimation module A, a signal demodulation module B and an amplitude limiting estimation noise reduction module C, an OQAM/FBMC symbol a is iteratively estimated through the linear estimation module A and the signal demodulation module B, and sparse distortion z is iteratively estimated through the amplitude limiting estimation noise reduction module C;
the specific method of iteration is as follows:
initializing iterative receiver parameters:wherein the numerical valueIs transmitted from the transmitting end to the receiving end,the method comprises the following steps that a mean value of z is represented, I represents an identity matrix, V represents a covariance matrix, an superscript pri is defined to represent prior information, a superscript post represents posterior information, a superscript ext represents external information, a subscript A represents a corresponding parameter input/output linear estimation module A, a subscript B represents a corresponding parameter input/output signal demodulation module B, a subscript C represents a corresponding parameter input/output amplitude limiting estimation noise reduction module C, a subscript k represents a kth symbol in a vector, and subscripts a and z of the covariance matrix V respectively represent covariance which is the symbol a and amplitude limiting distortion z; the signal processing mode of each module of the receiver is as follows:
s51, inputting the received signal into a linear estimation module A, passing the signal through an analysis filter bank before the linear estimation module A estimates a, firstly passing the received signal through a filter bank with the length of LgThe rectangular sliding window obtains a section of signal by moving M/2 sampling points, and the symbol estimation value is obtained by FFT transformation, matched filtering, phase rotation and real part operation
S52, combining the clipping noise and the channel noise into demodulation noise η ═ WHHz+WHn is and haveNote the bookTaking the real part of the output of the analysis filter bank, and then carrying out single-point LMMSE estimation, wherein k is (n-1) M + M, and then the estimated value of the kth symbol is as follows:
the MSE matrix calculation formula is:
wherein HmIs the channel frequency domain response;
inputting the external information into a signal demodulation module B;
s54, in the signal demodulation module B, firstly reconstructing QAM symbol from OQAM symbol, and makingThe reconstructed symbol is recorded asThen, the estimated value of the symbol s is optimized through soft demodulation, and each constellation point is assumed to contain q bits, so that the QAM modulation order is 2qThe symbol corresponding to the constellation point is marked as cl,l=1,2,...,2qThe log-likelihood ratio of the jth bit of the ith symbol output by the soft demodulator is recorded as lambdai,jI 1, 2., NM, j 1, 2.,. q, then the probability that the ith symbol corresponds to the ith constellation point is:
wherein b isi,jDenotes the ith symbolJ-th bit of cl,jRepresenting the ith constellation pointHas a probability of Pr (b) of 1i,j=1|λi,j)=exp(λi,j)/(1+exp(λi,j));
S55, corresponding to the OQAM symbol structure, separately calculating the real part and the imaginary part to obtain the symbol posterior information in the nth period as follows:
wherein 2(N-1) M +1 ≤ k ≤ 2nM, N ═ 1, 2.., N;
after passing through the signal demodulation module B, posterior information is obtainedAndwill posterior informationAndthe input linear estimation module A updates parameters:
s56, in the linear estimation module A, LMMSE estimation is carried out on the amplitude limiting distortion z:
the MSE matrix calculation formula is as follows,
s58, estimating in an amplitude-limiting estimation noise reduction module C based on amplitude-limiting distortion sparsity, and assuming that z obeys Gaussian Bernoulli distribution, namelyLambda is the clipping distortion sparsity and beta is the gaussian distribution variance.
wherein
The posterior variance is as follows,
And S510, if the algorithm is converged, ending, otherwise, returning to the step S52.
The invention has the beneficial effects that: estimating a truncated portion z of the signal while effectively reducing the PAPR of the signal to more accurately recover the signal; the invention carries out the Clipping and Filtering processing and uses the Turbo-CS algorithm to estimate the distorted signal in the time domain, on one hand, the PAPR of the signal is reduced by the Clipping at the transmitting end and the Filtering is used to relieve the side lobe regeneration, on the other hand, the Turbo-CS algorithm is used at the receiving end to detect the distorted signal z to ensure the performance of the error rate of the system.
Drawings
Fig. 1 is a diagram of a transmit-end modulation matrix structure;
FIG. 2 is a flow chart of a method of the present invention;
FIG. 3 is a graph showing the influence of the Clipping and Filtering performed by the transmitter on the PAPR;
FIG. 4 is a simulation curve of bit error rates of different processing modes in the OQAM/FBMC system.
Detailed Description
The present invention will be described in further detail with reference to the accompanying drawings.
As shown in fig. 2, a specific signal processing flow of the present invention includes repeated Clipping and Filtering at the transmitting end, introduction of channels, and Turbo-CS iterative receiver at the receiving end.
Setting the carrier number M used at the transmitting end to be 72, the number N of OQAM/FBMC symbols in a data frame to be 10, and the modulation mode is16-order QAM modulation with a symbol overlap factor K of 4, using a PHYDYAS project prototype filter with a filter length of 288 samples, definedWherein A is a threshold value, the amplitude limiting rate is set to be CR equal to 2dB, 1-time Clipping and Filtering processing is carried out at a transmitting end, and a simulation channel is an AWGN channel.
According to the setting of the parameters, the specific steps are as follows:
s1, inputting binary bit stream b [ k ]]QAM modulating to obtain a mapping symbol c with length of 72m,nAs an OQAM/FBMC symbol, two real symbols with the length of 72 are obtained by taking a real part and an imaginary partAndnoting that the nth transmitted symbol isThe transmitted symbol is subjected to phase rotation, frequency domain filtering and inverse fast fourier transform to obtain an up-sampled signal vector s of length 288n,
Sn=FHGΘnan
Wherein the filter matrix isGmFor the long 288 prototype filter frequency domain response on the m sub-carrier, phase rotatesθm,n=ejπ(m-n)/2ejnmπF is Lg×LgUnit DFT matrix, (.)HRepresenting a conjugate transpose.
S2, taking 10 OQAM/FBMC symbols as a data frame, delaying the generated 20 signal vectors by half a symbol period in turn, namely 36 sampling points, generating a transmitting signal after superposition,
s=Wa
whereinNote Wn=FHGΘnW is the modulation matrix, and L is the total length of the transmitted signals972. The s is subjected to Clipping and Filtering once and then transmitted to a transmitting antenna.
S3, the signal passes through AWGN channel, and the channel response matrix is the identity matrix I.
S4, the signal arrives at the receiving end to obtain the length LsThe observed signal of (a) is,
r=HWa+Hz+n
where H ═ I is the channel response matrix, z is the interference signal generated by the Clipping and filtering operation, it is necessary to estimate and cancel the interference at the receiving end and estimate the gaussian white noise with symbol a, n being zero mean.
S5, the receiving end cancels the interference and estimates the symbol a, specifically:
initializing iterative receiver parameters:wherein the numerical valueIs transmitted from the transmitting end to the receiving end,the mean value of z is represented, I represents the identity matrix, and we use the abbreviation "pri" with the superscript "prior" to represent prior information, the abbreviation "post" with the superscript "posterior" to represent posterior information, and the abbreviation "ext" with the superscript "externic" to represent extrinsic information.
S51, before module A estimates a, the signal is passed through an analysis filter bank, and the received signal is first passed through a filter with length Lg288 rectangular sliding window, taking a segment of signal every 36 sampling points, FFT transforming, matched filtering, phase rotating and real-taking operationTo obtain a symbol estimate
The matched filter coefficients are the conjugate of the frequency domain response function of the prototype filter, so the matrix form is
S52, combining the clipping noise and the channel noise into demodulation noise η ═ WHHz+WHn is and haveTaking the real part of the output of the analysis filter bank, and then carrying out single-point LMMSE estimation, wherein k is 72(n-1) + m, the estimated value of the kth symbol is,
the MSE matrix is calculated by the formula,
wherein HmIs the channel frequency domain response.
S53, the extrinsic information calculation formula is as follows,
S54, reconstructing QAM symbol from OQAM symbol in module BThe reconstructed symbol is recorded asThe estimate of the symbol s is then optimized by soft demodulation. Assuming that each constellation point contains 4 bits, so the QAM modulation order is 16, and the symbol corresponding to the constellation point is marked as c l1, 2.. 16. the log-likelihood ratio of the jth bit of the ith symbol output by the soft demodulator is denoted as λi,jI 1, 2, 720, j 1, 2, 4, the probability that the ith symbol corresponds to the ith constellation point is,
wherein b isi,jDenotes the ith symbolJ-th bit of cl,jRepresenting the ith constellation pointHas a probability of Pr (b) of 1i,j=1|λi,j)=exp(λi,j)/(1+exp(λi,j))。
S55, corresponding to OQAM symbol structure, calculating the real part and imaginary part separately to obtain the symbol posterior information in the nth period,
wherein 144(n-1) +1 ≦ k ≦ 144n, n ═ 1, 2
Using the posterior information of module B as input to module A, i.e.
S56, LMMSE estimation is carried out on the amplitude limiting distortion z in the module A, the calculation formula of the estimation value is as follows,
the MSE matrix calculation formula is as follows,
s57, the extrinsic information calculation formula is as follows,
the external information is transmitted to an amplitude limiting estimation noise reduction module C, and the order is
S58, estimating based on the clipping distortion sparsity in a module C, wherein the MMSE estimation calculation formula is as follows,
wherein
The posterior variance is as follows,
S59, an external information calculation formula,
And S510, if the algorithm is converged, ending, otherwise, returning to the step S52.
FIG. 3 is a graph showing the effect of Clipping and Filtering on the PAPR of a signal at a transmitter, where the abscissa indicates that the symbol power of dB-converted transmission is greater than the average power, and the ordinate showsThe ratio of the symbols is shown. It can be seen that 10 is taken from CCDF (complementary Current Distribution function)-5When the method is used, the PAPR is reduced by about 5dB after one-time Clipping and Filtering, and the obvious inhibition effect is achieved.
FIG. 4 shows the bit error rate performance of different systems under OQAM/FBMC system, where the bit error rate of the system is limited to BER 10-4And in time, the performance gain of about 3.6dB is obtained after one-time iterative compensation, and the performance gain of multiple-time iterative compensation is smaller. In addition, with the improvement of the signal to noise ratio, the difference of the performance curve of the algorithm of the invention and the performance curve of the algorithm without the Clipping operation is very small, and the PAPR is reduced while the detection performance is still very good.
Claims (1)
1. A signal transceiving method of an OQAM/FBMC system based on compressed sensing amplitude limiting is characterized by comprising the following steps:
s1, inputting binary bit stream b [ k ]]Obtaining a mapping symbol c with the length of M through QAM modulationm,nAs an OQAM/FBMC symbol, two real symbols with the length of M are obtained by taking a real part and an imaginary partAndnoting that the nth transmitted symbol isCarrying out phase rotation, frequency domain filtering and inverse fast Fourier transform on the transmitted symbol to obtain a length LgKM-up-sampled signal vector snAnd K is an overlapping factor and represents the overlapping number of OQAM/FBMC symbols:
sn=FHGΘnan
wherein the filter matrix isGmIs the m-th subcarrierLength L of wavelengthgThe prototype filter frequency domain response, phase rotationθm,n=ejπ(m+n)/2ejnmπF is Lg×LgUnit DFT matrix, (.)HRepresents a conjugate transpose;
s2, taking N OQAM/FBMC symbols as a data frame, delaying the generated 2N signal vectors by half a symbol period in sequence, namely M/2 sampling points, and generating a transmission signal after superposition:
s=Wa
whereinModulation matrix Wn=FHGΘnTotal length of transmitted signal isThe s is subjected to Clipping and Filtering processing and then is transmitted to a transmitting antenna;
s3, the signal is transmitted by the transmitting antenna, the signal passes through the multipath channel, because the tail value of the prototype filter is close to 0, the process is the process of the cyclic convolution of the signal and the channel;
s4, receiving the signal by the receiving end to obtain the length LsThe observed signals of (a) are:
r=HWa+Hz+n
wherein H is a channel response matrix, z is an interference signal generated by the Clipping and filtering operation, n is zero-mean Gaussian white noise, and the noise variance is
S5, the receiving end cancels the interference and estimates the symbol a, specifically: the receiver comprises a linear estimation module A, a signal demodulation module B and an amplitude limiting estimation noise reduction module C, an OQAM/FBMC symbol a is iteratively estimated through the linear estimation module A and the signal demodulation module B, and sparse distortion z is iteratively estimated through the amplitude limiting estimation noise reduction module C;
the specific method of iteration is as follows:
initializing iterative receiver parameters:wherein the numerical valueIs transmitted from the transmitting end to the receiving end,the method comprises the following steps that a mean value of z is represented, I represents an identity matrix, V represents a covariance matrix, an superscript pri is defined to represent prior information, a superscript post represents posterior information, a superscript ext represents external information, a subscript A represents a corresponding parameter input/output linear estimation module A, a subscript B represents a corresponding parameter input/output signal demodulation module B, a subscript C represents a corresponding parameter input/output amplitude limiting estimation noise reduction module C, a subscript k represents a kth symbol in a vector, and subscripts a and z of the covariance matrix V respectively represent covariance which is the symbol a and amplitude limiting distortion z; the signal processing mode of each module of the receiver is as follows:
s51, inputting the received signal into a linear estimation module A, passing the signal through an analysis filter bank before the linear estimation module A estimates a, firstly passing the received signal through a filter bank with the length of LgThe rectangular sliding window obtains a section of signal by moving M/2 sampling points, and the symbol estimation value is obtained by FFT transformation, matched filtering, phase rotation and real part operation
S52, combining the clipping noise and the channel noise into demodulation noise η ═ WHHz+WHn is and haveNote the bookTaking the real part of the output of the analysis filter bank, and then carrying out single-point LMMSE estimation, wherein k is (n-1) M + M, and then the estimated value of the kth symbol is as follows:
the MSE matrix calculation formula is:
wherein HmIs the channel frequency domain response;
inputting the external information into a signal demodulation module B;
s54, in the signal demodulation module B, firstly reconstructing QAM symbol from OQAM symbol, and makingThe reconstructed symbol is recorded asNM, then optimizing the estimate of the symbol s by soft demodulation, assuming that each constellation point contains q bits, so the QAM modulation order is 2qThe symbol corresponding to the constellation point is marked as cl,l=1,2,...,2qThe log-likelihood ratio of the jth bit of the ith symbol output by the soft demodulator is recorded as lambdai,jI 1, 2., NM, j 1, 2.,. q, then the probability that the ith symbol corresponds to the ith constellation point is:
wherein b isi,jDenotes the ith symbolJ-th bit of cl,jRepresenting the ith constellation pointHas a probability of Pr (b) of 1i,j=1|λi,j)=exp(λi,j)/(1+exp(λi,j));
S55, corresponding to the OQAM symbol structure, separately calculating the real part and the imaginary part to obtain the symbol posterior information in the nth period as follows:
wherein 2(N-1) M +1 ≤ k ≤ 2nM, N ═ 1, 2.., N;
after passing through the signal demodulation module B, posterior information is obtainedAndwill posterior informationAndthe input linear estimation module A updates parameters:
s56, in the linear estimation module A, LMMSE estimation is carried out on the amplitude limiting distortion z:
the MSE matrix calculation formula is as follows,
s58, estimating in an amplitude-limiting estimation noise reduction module C based on amplitude-limiting distortion sparsity, and assuming that z obeys Gaussian Bernoulli distribution, namelyLambda is amplitude limiting distortion sparsity, and beta is Gaussian distribution variance;
wherein beta is
The posterior variance is as follows,
And S510, if the algorithm is converged, ending, otherwise, returning to the step S52.
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