CN104901708A - Compressive sampling broadband digital receiver and signal processing method thereof - Google Patents

Compressive sampling broadband digital receiver and signal processing method thereof Download PDF

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CN104901708A
CN104901708A CN201510054075.2A CN201510054075A CN104901708A CN 104901708 A CN104901708 A CN 104901708A CN 201510054075 A CN201510054075 A CN 201510054075A CN 104901708 A CN104901708 A CN 104901708A
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陈涛
郭立民
赵忠凯
陈亚
刘志武
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Harbin Engineering University
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Abstract

The present invention discloses a compressive sampling broadband digital receiver and a signal processing method thereof. The broadband digital receiver includes an analog-to-digital converter, a mixer, a lowpass pre-filter, a decimator, a uniform channelization filter and a channel selection module. The analog-to-digital converter collects a signal x(n) according to a sampling frequency f<NYQ>. The mixer mixes the received signal with a pseudorandom sequence p(n) to obtain a mixed sequence x(n). The lowpass pre-filter filters the mixed sequence to obtain a filtered sequence x<D>(n). The decimator performs decimation by an integer factor R<D> on the filtered sequence to obtain a sequence x<p>(n). The uniform channelization filter processes the received sequence to obtain M output signals y<i><d>(n). The channel selection module selects, from the received signals, first R signals to obtain the final R output signals. The receiver of the present invention has a simple structure, can lower the system complexity, and can achieve the Nyquist sampling.

Description

Compression sampling broadband digital receiver and signal processing method thereof
Technical Field
The invention belongs to the field of data acquisition and processing, and particularly relates to a sub-Nyquist sampling broadband digital receiver for communication or radar signals and a signal processing method thereof.
Background
The broadband digital receiver has important application in the fields of spectrum sensing, space signal acquisition and identification. With the development of analog-to-digital conversion (ADC) acquisition technology, digital receivers are approaching to antennas more and more, and compared with conventional analog receivers, the digital receivers have significant advantages in the aspects of fully digital acquisition, post analysis and processing of signals, improvement of system sensitivity of the receivers, system reconfiguration and the like. At present, a uniform channelization structure, such as a WOLA structure, an FFT-based structure, and a CEM-based structure, is mostly adopted by a broadband digital receiver. Since the basic principle is based on constructing a uniform sub-channel structure by using an equal-bandwidth digital filter bank, the total sampling rate of the system is still the Nyquist (Nyquist) rate, and the design takes the setting of band-pass sampling and the setting of mixing into consideration.
Disclosure of Invention
The invention aims to provide a simple-structured broadband digital receiver applied to compressed sampling of sub-Nyquist sampling, and further aims to provide a signal processing method of the broadband digital receiver applied to compressed sampling, which can reduce the complexity of a system.
A compressed sampling broadband digital receiver comprises an analog-digital converter, a mixer, a pre-low pass filter, a decimator, a uniform channelization filter and a channel selection module,
analog-to-digital converter at sampling frequency fNYQCollecting signals x (n) and transmitting the signals x (n) to a mixer;
the mixer combines the received signal x (n) with a pseudo-random sequenceMixing to obtain a mixed sequenceTransmitting to a pre-low pass filter;
pre-low pass filter pair mixing post sequenceFiltering to obtain a filtered sequenceTransmitting to the extractor;
after the filter sequence of the decimator pairCarrying out RDObtaining the sequence after the multiple extractionTo a uniform channelizing filter;
the uniform channelized filter processes the received sequence to obtain M paths of output signalsTransmitting to the channel selection module;
the channel selection module receives the signal from the received signalThe front R path signal is selected to obtain the final R path output signal yj(n),j=0,1,2...R-1。
A signal processing method for a compressed-sample wideband digital receiver, comprising the steps of:
the method comprises the following steps: analog-to-digital converter at sampling frequency fNYQCollecting signals x (n) and transmitting the signals x (n) to a mixer;
step two: the mixer combines the signal x (n) with a pseudo-random sequenceThe frequency-mixing process is carried out,r ∈ 1, Z is an integer, Mp=Tp·fNYQ,TpIs the time period of the pseudorandom sequence, let frequencyB is the input signal bandwidth;
step three: pre-low pass filter pair mixing post sequenceFiltering to obtain a filtered sequenceThe frequency-domain values of the filtered sequence are:
<math><mrow> <msub> <mover> <mi>x</mi> <mo>~</mo> </mover> <mi>D</mi> </msub> <mrow> <mo>(</mo> <msup> <mi>e</mi> <mi>j&omega;</mi> </msup> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <mn>1</mn> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <munderover> <mi>&Sigma;</mi> <mrow> <mi>k</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>P</mi> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <mi>X</mi> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>-</mo> <mi>k</mi> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>)</mo> </mrow> <mo>,</mo> <mo>-</mo> <mfrac> <mi>&pi;</mi> <msub> <mi>R</mi> <mi>D</mi> </msub> </mfrac> <mo>&le;</mo> <mi>&omega;</mi> <mo>&le;</mo> <mfrac> <mi>&pi;</mi> <msub> <mi>R</mi> <mi>D</mi> </msub> </mfrac> </mrow></math>
wherein R isDTake not more than fNYQ/(2(Rfp+fT) Maximum integer of);
step four: after the filtering sequence passes through the decimator, every RDSampling a sampling point once to obtain a sequence
Step five: the uniform channelized filter processes the received sequence to obtain M paths of output signals
y i d ( n ) , i = 0,1,2 . . . M - 1 ;
Step six: the channel selection module receives the signal from the received signalThe front R path signal is selected to obtain the final R path output signal yj(n, j ═ 0,1,2.. R-1, and the sampling frequency of each channel is fs=fNYQ/(RDK) K is the decimation rate of the M uniform channelized filters, fs=fpAnd is and
has the advantages that:
(1) a broadband digital receiver is constructed by adopting MWC theory. The MWC theory is mainly used for applications of Analog Information Conversion (AIC). The invention is applied to the realization of a broadband digital receiver, on one hand, the sub-Nyquist sampling can be ensured to be realized, and on the premise of ensuring that the input signal information can be completely recovered, less data is adopted for storage, thereby saving the storage resource of the receiver. On the other hand, because the frequency mixing modulation is carried out by adopting the pseudo-random signal, the structure of the invention can be used on the premise of ensuring the Nyquist sampling of the input signal without considering the arrangement problem of band-pass sampling and frequency mixing in the traditional broadband digital receiving design.
(2) A low pass filter and a decimator. The invention arranges a low-pass filter and a decimator between the mixer and the channelized receiver, and aims to reduce the number of paths arranged by the uniform channelized receiver so as to reduce the complexity of the system. The decimator is operative to perform a first down sampling of the low-pass filtered signal.
(3) The single-pass mixed MWC architecture is combined with a uniform channelized receiver architecture. The invention adopts the MWC structure of single-path mixing design, considers the spectrum characteristic of random signal modulation, namely the mixed signalThe DTFT of (1) is x (e) of the DTFT of the input signal x (n)) With fpIs a linear combination of periodic shifts. As shown in fig. 4, the frequency spectrum is fpAre evenly spaced. A uniform channelized receiver structure can be further employed. This allows the decimation factor to be placed at the front end of the channelized receiver, reducing the operating frequency of the system, thereby further reducing the complexity of the system design.
The invention relates to a broadband digital receiver design capable of realizing sub-Nyquist compression sampling under the condition of ensuring the restorable signal. A wideband digital receiver typically receives one or more known or unknown signals over a period of time. This way of receiving the signal is consistent with a multi-band signal model, i.e. a frequency domain sparse signal. Therefore, according to the compressed sensing principle, for the spectrum sparse signals, the analog-digital conversion can be expanded to the Analog Information Conversion (AIC), so that redundant information is removed, and the sampled data can be further compressed. A modulation broadband conversion (MWC) model based on compressive sampling can well process multi-band signals, and is a successful AIC compressive sampling structure. MWC theory requires that the signal be mixed with multiple pseudo-random (PN) sequences. The selection of the pseudo-random sequence has a great influence on the reconstruction of the signal. In order to reduce the complexity of PN sequence selection, the invention adopts a single-path mixing model, and relies on the periodic shift of the signal generated by pseudo-random signal modulation in a frequency domain, thereby constructing a plurality of sub-band filter filters and realizing the compression sampling of the signal. The single-path mixing model is very suitable for constructing M paths of uniform channelized receivers to realize, so that R paths of uniform channelized receivers are selected from the M paths of uniform channelized receivers to serve as observation matrixes for compression sampling, and finally original signals can be recovered from R paths of data by adopting a CTF algorithm.
Drawings
FIG. 1 is a functional block diagram of the wideband digital receiver of the present invention;
FIG. 2 is a schematic block diagram of a uniform channelization filter based on a polyphase structure;
fig. 3 is a spectrogram illustration of a multi-band signal (N ═ 2) and a pseudorandom sequence;
FIG. 4 is a schematic of a spectrogram after mixing a pseudorandom sequence with an input signal;
FIG. 5 is a 6-way compressed sample of data of a Linear Frequency Modulated (LFM) signal;
fig. 6 is a graph of the frequency domain comparison of the reconstructed signal of the LFM signal with the original signal;
fig. 7 is a comparison graph of the reconstructed signal of the LFM signal with the original signal in the time domain;
figure 8 MSE performance of the wideband digital receiver of the present invention is compared to the standard MWC algorithm.
Detailed Description
The present invention will be described in further detail below with reference to the accompanying drawings.
The invention is applied to digital receiving systems such as communication, radar and the like, and aims to provide an effective sub-Nyquist sampling broadband digital receiver structure which can ensure accurate recovery of original signals.
The purpose of the invention is realized as follows: via an analog-to-digital converter at a sampling frequency fNYQThe collected signal x (n) is first of all related to the period TpPseudo-random sequence ofMixing to obtain a mixed sequenceThen theWith a cut-off frequency of [0, fNYQ/2RD]H ofD(n) pre-low pass filter filtering, followed by RDObtaining the sequence after the multiple extraction The sequence enters an M-path uniform channelized receiver; and finally, selecting the front R paths from the M paths of outputs as the observation data of the compressed samples.
The invention is applicable to the following conditions:
(1) the input signal can be a multi-band signal or a broadband radar signal (such as an LFM signal), and the final number R of the selected paths is required to be more than or equal to 4N if the input signal has N signals with the maximum bandwidth not exceeding B.
(2) x (n) the sampling sequence is set to satisfy the Nyquist sampling theorem and has a sampling frequency of fNYQ
(3)Is one period of MpPeriodic pseudo-random sequences of, i.e.r ∈ 1, where 1 is an integer. Mp=Tp·fNYQ,TpIs the time period of the pseudo-random sequence,is p (n), and p (n) can be completed by a binary +/-1 Bernoulli random sequence. Require that
(4) Compressing the sampled R-path sampled data y1(n),y2(n)…,yr(n)…,yR(n), the sampling frequency of each path is fs=fNYQ/(RDK) And K is the extraction rate of the M paths of uniform channelized receivers. Requirement fs≥fpIn the present invention, f is takens=fp
From this weThe total sampling rate of the system is obtained asGetA sub-nyquist sampling system can be obtained.
According to the MWC principle of compression sampling, M paths of uniform channelized filtering are carried out on sequences corresponding to periodically shifted linear combined frequency spectrums obtained by frequency mixing with pseudo-random sequences, and then R paths of uniform channelized filtering are selected from the sequences to serve as compression sampling data. Meanwhile, in order to reduce the number of channelized paths and the complexity of a system, a stage of pre-low-pass filtering and decimation operation is added between the frequency mixer and the channelized filtering. The invention is described in detail below with reference to the figures and examples.
Referring to fig. 1, a schematic block diagram of a system according to the present invention is shown, the system is composed of a digital mixing module, a low pass filter module, a decimator module, a channelization filter module, and a channel selection module, wherein the channelization filter module is specifically implemented as shown in fig. 2. E in FIG. 2l(z) is the l-th polynomial component of the prototype filter in the channelization design, and the IDFT represents the inverse discrete fourier transform. In the following we analyze the processing flow of the signals in turn.
The discrete-time Fourier transform (DTFT) of the input signal x (n) is
<math><mrow> <mi>x</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>-</mo> <mfrac> <mn>1</mn> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> </mfrac> <msubsup> <mo>&Integral;</mo> <mrow> <mo>-</mo> <mi>&pi;</mi> </mrow> <mi>&pi;</mi> </msubsup> <mi>X</mi> <mrow> <mo>(</mo> <msup> <mi>e</mi> <mi>j&omega;</mi> </msup> <mo>)</mo> </mrow> <msup> <mi>e</mi> <mi>jn&omega;</mi> </msup> <mi>d&omega;</mi> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>1</mn> <mo>)</mo> </mrow> </mrow></math>
Here, ,ω=2πfT,fNYQ1/T is the Nyquist sampling frequency. The pseudo-random is a Bernoulli sequence taking { -1, +1} binary valuesHas a discrete Fourier series of
<math><mrow> <mover> <mi>p</mi> <mo>~</mo> </mover> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <mn>1</mn> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <munderover> <mi>&Sigma;</mi> <mrow> <mi>k</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>P</mi> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <msup> <mi>e</mi> <mrow> <mi>j</mi> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <mi>N</mi> </mfrac> <mi>nk</mi> </mrow> </msup> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>2</mn> <mo>)</mo> </mrow> </mrow></math>
Where P (k) is a pseudo-random sequenceOf main value sequences p (n)And (4) Fourier transform. Thus the output signal after the mixer x ~ ( n ) = x ( n ) p ~ ( n ) DTFT conversion of
<math><mrow> <mover> <mi>x</mi> <mo>~</mo> </mover> <mrow> <mo>(</mo> <msup> <mi>e</mi> <mi>j&omega;</mi> </msup> <mo>)</mo> </mrow> <mo>=</mo> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mo>-</mo> <mo>&infin;</mo> </mrow> <mo>&infin;</mo> </munderover> <mi>x</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>&CenterDot;</mo> <mover> <mi>p</mi> <mo>~</mo> </mover> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <mi>e</mi> <mrow> <mo>-</mo> <mi>jn&omega;</mi> </mrow> </msup> <mo>=</mo> <mfrac> <mn>1</mn> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <munderover> <mi>&Sigma;</mi> <mrow> <mi>k</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>P</mi> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <mi>X</mi> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>-</mo> <mi>k</mi> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>)</mo> </mrow> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>3</mn> <mo>)</mo> </mrow> </mrow></math>
Here, the present invention takes N ═ 2 as an example, and fig. 3 shows a schematic of DTFT transform of an input multiband signal (N ═ 2) and fourier series of a pseudo-random sequence; figure 4 gives a spectral representation of the pseudorandom sequence mixed with the input multi-band signal.
ByAs can be seen from the DTFT conversion result formula (3) and FIG. 4, the DTFT conversion resultsDTFT transform of x (n) x (e)) At a frequency fpLinear combination of periodic shifts whose frequency spectrum is at fpAre evenly spaced. And therefore the R subbands among them may be selected to constitute the compressively sampled observation data. To achieve filtering for the R subband, the present invention selects a uniform polyphase structure based channelized filter receiver as shown in fig. 2. However, if the structure of FIG. 2 is directly adopted, the system will followThe number of sub-bands is increased, the number of channels is increased, so that the system becomes complex, and the hardware resource occupation has higher requirements. The passband of the pre-low pass filter is RfpThe bandwidth of each sub-band may be channelized to be fp. Meanwhile, in order to reduce the design difficulty of the front low-pass filter and the design order, the transition band of the low-pass filter is optionally wider (set as f)T) Then the total bandwidth of channelization is designed to be Rfp+fT. Thus is equivalent toThrough R Hr(f) Sub-filters, Hr(f) Is defined as:
<math><mrow> <msub> <mi>H</mi> <mi>r</mi> </msub> <mrow> <mo>(</mo> <mi>f</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfenced open='{' close=''> <mtable> <mtr> <mtd> <mn>1</mn> </mtd> <mtd> <mrow> <mo>(</mo> <mn>2</mn> <mi>r</mi> <mo>-</mo> <mn>3</mn> <mo>)</mo> </mrow> <mfrac> <msub> <mi>f</mi> <mi>s</mi> </msub> <mn>2</mn> </mfrac> <mo>&le;</mo> <mi>f</mi> <mo>&le;</mo> <mrow> <mo>(</mo> <mn>2</mn> <mi>r</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mfrac> <msub> <mi>f</mi> <mi>s</mi> </msub> <mn>2</mn> </mfrac> <mo>,</mo> <mn>1</mn> <mo>&le;</mo> <mi>r</mi> <mo>&le;</mo> <mi>R</mi> </mtd> </mtr> <mtr> <mtd> <mn>0</mn> </mtd> <mtd> <mi>otherwise</mi> </mtd> </mtr> </mtable> </mfenced> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>4</mn> <mo>)</mo> </mrow> </mrow></math>
in summary, thenAfter passing through the low-pass filtering module, the signal isIts frequency domain output is:
<math><mrow> <msub> <mover> <mi>x</mi> <mo>~</mo> </mover> <mi>D</mi> </msub> <mrow> <mo>(</mo> <msup> <mi>e</mi> <mi>j&omega;</mi> </msup> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <mn>1</mn> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <munderover> <mi>&Sigma;</mi> <mrow> <mi>k</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>P</mi> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <mi>X</mi> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>-</mo> <mi>k</mi> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>)</mo> </mrow> <mo>,</mo> <mo>-</mo> <mfrac> <mi>&pi;</mi> <msub> <mi>R</mi> <mi>D</mi> </msub> </mfrac> <mo>&le;</mo> <mi>&omega;</mi> <mo>&le;</mo> <mfrac> <mi>&pi;</mi> <msub> <mi>R</mi> <mi>D</mi> </msub> </mfrac> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>5</mn> <mo>)</mo> </mrow> </mrow></math>
where R isDTake not more than fNYQ/(2(Rfp+fT) Maximum integer of) signalWarp RDAfter the decimator module, the time domain signalEvery RDA signal obtained by sampling a sampling point onceThe spectrum is spread to-pi ≦ ω ≦ pi. Then passes through a channelized filter module to output y from the mth subchannel of the systemd m(n) DTFT conversion of the signal Yd m(ω) is:
<math><mrow> <msub> <msup> <mi>Y</mi> <mi>d</mi> </msup> <mi>m</mi> </msub> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>)</mo> </mrow> <mo>=</mo> <munderover> <mi>&Sigma;</mi> <mrow> <mi>l</mi> <mo>=</mo> <mo>-</mo> <msub> <mi>L</mi> <mn>0</mn> </msub> </mrow> <msub> <mi>L</mi> <mn>0</mn> </msub> </munderover> <msup> <mi>P</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mi>l</mi> <mo>)</mo> </mrow> <mi>X</mi> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>-</mo> <mrow> <mo>(</mo> <mi>l</mi> <mo>-</mo> <mi>m</mi> <mo>)</mo> </mrow> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>)</mo> </mrow> <mo>,</mo> <mi>&omega;</mi> <mo>&Element;</mo> <mo>[</mo> <mo>-</mo> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>,</mo> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>]</mo> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>6</mn> <mo>)</mo> </mrow> </mrow></math>
finally via R-channel selection modules, i.e. from channelisation filtersToThe front R paths are selected from the M paths of output to be used as the output of compression sampling, and the matrix form is written into by considering all R selected branches
Y(ω)=Φz(ω) (7)
Wherein <math><mrow> <munder> <mi>z</mi> <mo>&OverBar;</mo> </munder> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>)</mo> </mrow> <mo>=</mo> <msup> <mfenced open='[' close=']'> <mtable> <mtr> <mtd> <mi>X</mi> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>+</mo> <msub> <mi>L</mi> <mn>0</mn> </msub> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>)</mo> </mrow> </mtd> <mtd> <mi>X</mi> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>+</mo> <mrow> <mo>(</mo> <msub> <mi>L</mi> <mn>0</mn> </msub> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>)</mo> </mrow> </mtd> <mtd> <mo>.</mo> <mo>.</mo> <mo>.</mo> </mtd> <mtd> <mn>0</mn> </mtd> <mtd> <mo>.</mo> <mo>.</mo> <mo>.</mo> </mtd> <mtd> <mi>X</mi> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>-</mo> <msub> <mi>L</mi> <mn>0</mn> </msub> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>)</mo> </mrow> </mtd> </mtr> </mtable> </mfenced> <mi>T</mi> </msup> </mrow></math> Is 2L in length0A column vector of +1, which is the sparse solution to be solved. <math><mrow> <mi>&Phi;</mi> <mo>=</mo> <msup> <mfenced open='[' close=']'> <mtable> <mtr> <mtd> <msubsup> <munder> <mi>&phi;</mi> <mo>&OverBar;</mo> </munder> <mn>1</mn> <mi>T</mi> </msubsup> </mtd> <mtd> <msubsup> <munder> <mi>&phi;</mi> <mo>&OverBar;</mo> </munder> <mn>2</mn> <mi>T</mi> </msubsup> </mtd> <mtd> <mo>.</mo> <mo>.</mo> <mo>.</mo> </mtd> <mtd> <msubsup> <munder> <mi>&phi;</mi> <mo>&OverBar;</mo> </munder> <mi>R</mi> <mi>T</mi> </msubsup> </mtd> </mtr> </mtable> </mfenced> <mi>T</mi> </msup> </mrow></math> Is of size R × (2L)0+1) compressed sensing matrix, wherein <math><mrow> <msubsup> <munder> <mi>&phi;</mi> <mo>&OverBar;</mo> </munder> <mi>r</mi> <mi>T</mi> </msubsup> <mo>=</mo> <mfenced open='[' close=']'> <mtable> <mtr> <mtd> <msubsup> <munder> <mn>0</mn> <mo>&OverBar;</mo> </munder> <mrow> <mi>r</mi> <mo>-</mo> <mn>1</mn> </mrow> <mi>T</mi> </msubsup> </mtd> <mtd> <msup> <mi>P</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mo>-</mo> <msub> <mi>L</mi> <mn>0</mn> </msub> <mo>)</mo> </mrow> </mtd> <mtd> <mo>.</mo> <mo>.</mo> <mo>.</mo> </mtd> <mtd> <msup> <mi>P</mi> <mo>&prime;</mo> </msup> </mtd> </mtr> </mtable> <mrow> <mrow> <mo>(</mo> <msub> <mi>L</mi> <mn>0</mn> </msub> <mo>-</mo> <mi>r</mi> <mo>+</mo> <mn>1</mn> <mo>)</mo> </mrow> <mi></mi> </mrow> </mfenced> <mo>,</mo> </mrow></math> L0Is chosen as the smallest integer that can include all non-zero values of the X (ω) spectrum.Is a column vector with m-1 zero elements.
<math><mrow> <msup> <mi>P</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mi>l</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfenced open='{' close=''> <mtable> <mtr> <mtd> <mfrac> <mn>1</mn> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mi>P</mi> <mrow> <mo>(</mo> <mi>l</mi> <mo>)</mo> </mrow> </mtd> <mtd> <mrow> <mo>(</mo> <mn>0</mn> <mo>&le;</mo> <mi>l</mi> <mo>&le;</mo> <msub> <mi>L</mi> <mn>0</mn> </msub> <mo>)</mo> </mrow> </mtd> </mtr> <mtr> <mtd> <mfrac> <mn>1</mn> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mi>P</mi> <mrow> <mo>(</mo> <mi>M</mi> <mo>+</mo> <mi>l</mi> <mo>)</mo> </mrow> </mtd> <mtd> <mrow> <mo>(</mo> <mo>-</mo> <msub> <mi>L</mi> <mn>0</mn> </msub> <mo>&le;</mo> <mi>l</mi> <mo><</mo> <mn>0</mn> <mo>)</mo> </mrow> </mtd> </mtr> </mtable> </mfenced> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>8</mn> <mo>)</mo> </mrow> </mrow></math>
Matrix arrayY(ω)=ΦzAnd (omega) solving is a solving problem of a sparse theory, and considering that a processed signal is a multi-band model, the solving problem is an IMV problem. Therefore, the solution can adopt a CTF algorithm to convert the IMV into the MMV problem.
From this we derive the total sampling rate of the system asGetA sub-nyquist sampling system can be obtained.
Typical simulation results are given below to verify the feasibility of the invention. Input signal having a frequency fNYQ960MHz, the number of sampling points is 4096 points; f. ofs=fp30 MHz; r ═ 6; the pre-bandpass filter is designed to have a cutoff frequency of 240 MHz. Fig. 5 shows 6 channels of compressed sample data after the input signal is a compressed sample of a chirp (LFM) signal. Fig. 6 and 7 show the frequency domain and time domain contrast diagrams of the LFM signal after compression sampling and the reconstructed signal and the original signal, respectively. The bandwidth of the LFM signal is 10MHz and the SNR is 20 dB. The sampling rate after compression sampling is R.fs180MHz and 960MHz, which is less than the nyquist sampling frequency. FIG. 7 shows the MSE performance of recovered signals at different SNR values from 0dB to 30dBAnd compared with the standard MWC algorithm. The MSE calculation formula is
MSE = | | x ^ ( n ) - x ( n ) | | 2 | | x ( n ) | | 2 - - - ( 9 )
Wherein,and x (n) are the reconstructed signal and the original signal, respectively. Because the invention only adopts one-way mixing, the MSE performance of the invention is better than that of the standard MWC algorithm. From the simulation, it can be seen that, as shown in fig. 8, the structure of the present invention can realize reconstruction of LFM and other signals received in a broadband under the condition of sub-nyquist sampling.

Claims (2)

1. A wideband digital receiver for compressing samples, characterized by: comprises an analog-digital converter, a frequency mixer, a pre-low pass filter, a decimator, a uniform channelizing filter and a channel selection module,
analog-to-digital converter at sampling frequency fNYQCollecting signals x (n) and transmitting the signals x (n) to a mixer;
the mixer combines the received signal x (n) with a pseudo-random sequenceMixing to obtain a mixed sequenceTransmitting to a pre-low pass filter;
pre-low pass filter pair mixing post sequenceFiltering to obtain a filtered sequenceTransmitting to the extractor;
after the filter sequence of the decimator pairCarrying out RDObtaining the sequence after the multiple extractionTo a uniform channelizing filter;
the uniform channelized filter processes the received sequence to obtain M paths of output signalsM-1, which is transmitted to the channel selection module;
the channel selection module receives the signal from the received signalThe front R path signal is selected to obtain the final R path output signal yj(n),j=0,1,2...R-1。
2. A method for signal processing in a compressed-sample wideband digital receiver according to claim 1, comprising the steps of:
the method comprises the following steps: analog-to-digital converter at sampling frequency fNYQThe collected signals x (n) being passed to the mixA frequency converter;
step two: the mixer combines the signal x (n) with a pseudo-random sequenceThe frequency-mixing process is carried out,r ∈ 1, Z is an integer, Mp=Tp·fNYQ,TpIs the time period of the pseudorandom sequence, let frequencyB is the input signal bandwidth;
step three: pre-low pass filter pair mixing post sequenceFiltering to obtain a filtered sequenceThe frequency-domain values of the filtered sequence are:
<math> <mrow> <msub> <mover> <mi>x</mi> <mo>~</mo> </mover> <mi>D</mi> </msub> <mrow> <mo>(</mo> <msup> <mi>e</mi> <mi>j&omega;</mi> </msup> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <mn>1</mn> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <munderover> <mi>&Sigma;</mi> <mrow> <mi>k</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>P</mi> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <mi>X</mi> <mrow> <mo>(</mo> <mi>&omega;</mi> <mo>-</mo> <mi>k</mi> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <msub> <mi>M</mi> <mi>p</mi> </msub> </mfrac> <mo>)</mo> </mrow> <mo>,</mo> <mo>-</mo> <mfrac> <mi>&pi;</mi> <msub> <mi>R</mi> <mi>D</mi> </msub> </mfrac> <mo>&le;</mo> <mi>&omega;</mi> <mo>&le;</mo> <mfrac> <mi>&pi;</mi> <msub> <mi>R</mi> <mi>D</mi> </msub> </mfrac> </mrow> </math>
wherein R isDTake not more than fNYQ/(2(Rfp+fT) Maximum integer of);
step four: after the filtering sequence passes through the decimator, every RDSampling a sampling point once to obtain a sequence
Step five: the uniform channelized filter processes the received sequence to obtain M paths of output signals
i=0,1,2...M-1;
Step six: the channel selection module receives the signal from the received signalThe front R path signal is selected to obtain the final R path output signal yj(n, j ═ 0,1,2.. R-1, and the sampling frequency of each channel is fs=fNYQ/(RDK) K is the decimation rate of the M uniform channelized filters, fs=fpAnd R is satisfied
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