The present application claims the benefit of the earlier filing date of EP11169553.2 filed in the European Patent Office on Jun. 10, 2011, the entire content of which application is incorporated herein by reference.
1. Field of the Disclosure
The present disclosure relates to a signal processing unit and method for accurately estimating from a time-domain signal at least one frequency and corresponding amplitude of at least one complex exponential. The present disclosure also relates to an object detection system for detecting at least one target object at a range, the system comprising such signal processing unit. The present disclosure further relates to a computer program and a computer readable non-transitory medium for implementing such a method.
2. Description of Related Art
An object detection system for determining a range of a target object is also known as a radar system (radio detection and ranging). A frequency modulated continuous wave (FMCW) object detection or radar system uses a transmission signal with linearly increasing frequency comprising a number of consecutive chirps. Such a system comprises a transmitter for transmitting a transmission signal and a receiver for receiving transmission signal reflections from the target object as a reception signal. The range of the target object can be estimated using the frequency difference between the transmission signal and the reception signal, also known a beat frequency. The beat frequency is directly proportional to the range of the target object.
The range resolution is defined as the minimum distance between two target objects which can be successfully separated. When using traditional frequency-domain techniques, this range resolution depends only on the bandwidth of the chirp and the speed of propagation. US 2009/0224978 A1 for example discloses a detection device and method using such frequency-domain techniques for direction of arrival estimation. In particular, US 2009/0224978 A1 discloses a detection device with a transmission sensor array and a reception sensor array that is formed of n sensor elements, estimating a target count indicating the number of targets based on reflected signals of transmission signals sent from the transmission sensor array and reflected from the targets, and estimating an angle at which each reflected signal comes based on the target count.
Time-domain techniques however allow finer resolution to be obtained (so-called “super-resolution” techniques). For example, U.S. Pat. No. 6,529,794 B1 discloses a FMCW sensor system. The emitted signals are received after reflection at targets and processed to form a measured signal whose frequency spectrum is analyzed. Discrete equidistant samples are arranged in a double Hankel matrix in the existing sequence. This matrix is diagonalized with a singular value decomposition and an approximation is identified taking only the principle values into consideration, in order to calculate the frequencies and their amplitudes there from using known methods.
The “background” description provided herein is for the purpose of generally presenting the context of the disclosure. Work of the presently named inventor(s), to the extent it is described in this background section, as well as aspects of the description which may not otherwise qualify as prior art at the time of filing, are neither expressly or impliedly admitted as prior art against the present invention.
It is an object to provide a signal processing unit and method for accurately estimating from a time-domain signal at least one frequency and corresponding amplitude for at least one complex exponential having improved performance, in particular providing less computational load, finer range resolution, improved robustness of low signal-to-noise (SNR) situations and/or reduced absolute and relative positioning errors. It is a further object to provide an object detection system comprising or using such signal processing unit, and a computer program and a computer readable non-transitory medium for implementing such signal processing method.
According to an aspect there is provided a signal processing unit for accurately estimating from a time-domain signal at least one frequency and corresponding amplitude of at least one complex exponential. The signal processing unit is configured to transform the time-domain signal into a frequency-domain signal and to detect at least one peak frequency in a power spectrum of the frequency-domain signal. The signal processing unit is further configured to determine at least one frequency band of interest corresponding to the at least one peak frequency and to determine at least one signal-to-noise ratio corresponding to the at least one frequency band of interest. Further, the signal processing unit is configured to perform at least one subsequent time-domain processing step for accurately estimating, based on the time-domain signal, the at least one frequency and corresponding amplitude of the at least one complex exponential using the at least one frequency band of interest and/or the at least one signal-to-noise ratio.
According to a further aspect there is provided an object detection system for detecting at least one target object at a range. The system comprises a transmitter for transmitting a transmission signal, and a receiver for receiving transmission signal reflections from the at least one target object as a reception signal. The receiver comprises a mixer for generating a mixed signal based on the transmission signal and the reception signal. The system further comprises the signal processing unitdisclosed herein, wherein the mixed signal is the time-domain signal.
According to a further aspect there is provided a signal processing method for accurately estimating from a time-domain signal at least one frequency and corresponding amplitude of at least one complex exponential. The method comprises transforming the time-domain signal into a frequency-domain signal and detecting at least one peak in a power spectrum of the frequency-domain signal. The method further comprises determining at least one frequency band of interest corresponding to the at least one peak and determining at least one signal-to-noise ratio corresponding to the at least one frequency band of interest. Further, the method comprises performing least one subsequent time-domain processing step for accurately estimating, based on the time-domain signal, at the at least one frequency and corresponding amplitude of the at least one complex exponential using the at least one frequency band of interest and/or the at least one signal-to-noise ratio.
According to still further aspects a computer program comprising program means for causing a computer to carry out the steps of the method disclosed herein, when said computer program is carried out on a computer, as well as a non-transitory computer-readable recording medium that stores therein a computer program product, which, when executed by a processor, causes the method disclosed herein to be performed are provided.
Preferred embodiments are defined in the dependent claims. It shall be understood that the claimed method, the claimed computer program and the claimed computer readable medium have similar and/or identical preferred embodiments as the claimed signal processing unit or object detection system and as defined in the dependent claims.
One of the aspects of the present disclosure is to perform a pre-estimation in an initial stage by determining at least one frequency band of interest and corresponding signal-to-noise ratio in the frequency-domain and to use the determined frequency band of interest or the corresponding signal-to-noise ratio (in particular the signal-to-noise ratio) in at least one (in particular multiple) subsequent time-domain processing step, in particular for super-resolution time-domain processing. In this way, further stages or steps can be adapted to the current situation. In particular, the computational load required in subsequent stages or steps can be optimally tuned. Just as an example, the determined frequency band of interest can be used in an adaptive band-pass filter for filtering the time-domain signal. In particular the determined signal-to-noise ration can be used in various ways. Just as an example, the determined signal-to-noise ration can be compared to a given threshold. In one example, subsequent processing steps can only be performed if the determined signal-to-noise ratio is equal to or above the given threshold. In another example, a specific method can be performed if the determined signal-to-noise ratio is equal or above the given threshold, and/or another method can be performed if the determined signal-to-noise threshold is below the given threshold. In another example, the determined signal-to-noise parameter can be used to define a certain parameter.
For example, when the determined signal-to-noise ratio from the preestimation stage is used in a model-order-selection algorithm for estimating a model-order, an improved robustness in low SNR situations or conditions can be achieved. In another example, when the determined signal-to-noise ratio from the pre-estimation stage is used in a reduced-rank-Hankel approximation (RRHA), an improved robustness in low SNR situations or conditions can be achieved. In a further example, when the determined signal-to-noise ratio from the pre-estimation stage is used in a refinement stage to refine the at least one estimated frequency, the absolute and relative positioning errors can be reduced.
With the signal processing unit and method using super-resolution time-domain processing described herein, a significantly improved result can be achieved compared to using conventional frequency-domain techniques, in particular an improved range resolution. For example, with the signal processing unit and method described herein, an improvement of the range resolution of up to 15 times can be achieved compared to a frequency-domain technique where the range resolution is limited to c/(2Δ,f).
It is to be understood that both the foregoing general description of the invention and the following detailed description are exemplary, but are not restrictive, of the invention.
A more complete appreciation of the disclosure and many of the attendant advantages thereof will be readily obtained as the same becomes better understood by reference to the following detailed description when considered in connection with the accompanying drawings, wherein:
It is now referred to the drawings, wherein like reference numerals designate identical or corresponding parts throughout the several views. In the following the signal processing unit and method will be described with reference to an object detection system, in particular the object detection system 10 according to the embodiment of
In the embodiment of
The object detection system 10 comprises a signal generator 11 for generating the FMCW transmission signal Tx. The objection detection system 10 further comprises a mixer 13 for generating a mixed signal based on the transmission signal Tx and the reception signal Rx. In the embodiment of
In general, a transmitter of the object detection system 10 can comprise the signal generator 11 and/or the transmitter antenna 15. A receiver of the object detection system 10 can comprise the receiver antenna 16 and/or the mixer 13. The output of the mixer 13 (mixed signal) is supplied to the signal processing unit 12. The transmitter and/or receiver can optionally further comprise analog filter(s) and/or amplifier(s). In
Apart from the beat frequency, a Doppler frequency is also present at the output of the mixer. Doppler effect occurs when the wave front of the transmission signal Tx transmitted by the system reaches the target object 1. This produces a frequency shift which is proportional to the velocity v of the target and to the central frequency fc. The following equation shows the dependence of on the beat frequency fbeat from range R, velocity v, transmitted chirp bandwidth Δf, chirp duration Tb, chirp center frequency fc and the speed of the light c:
The time-domain (or mixed) signal is generally first processed in the frequency-domain, by using a frequency transformation operator. The frequency transformation operator can in particular be a Fast Fourier Transformation (FFT). The following equation shows an example of an analytical expression of a transformation of the time-domain signal, produced by a number of consecutive chirps, to the frequency-domain using FFT:
Y(f, k)=∫(k−1/2)·Tb+T
where y(t) is the time-domain signal, Tb is the duration of one chirp, f is frequency, t is time and k is the number of chirps.
Then a power spectrum is determined in the frequency-domain and a peak detection method is performed. There exist multiple known methods to perform peak detection. One example is thresholding, wherein a target object is detected when the power in the power spectrum overcomes a certain value. Another example is CFAR (Constant False Alarm Rate) wherein the detection is adapted dynamically. However, it shall be understood that any other suitable peak detection method can be used.
The two main factors that may limit the detection of the range R of a target object are the maximum range Rmax that can be detected and the range resolution δR. The maximum detectable range Rmax is limited by the duration Tb of the chirp used. The chirp duration Tb has to be much larger (for example ten times or more) than the flight time Tp. Another limiting factor is the maximum bandwidth of the signal which is for example half (or lower) than the sampling rate of the sampling unit.
When the detection is performed in the frequency-domain, the range resolution depends on the bandwidth of the used chirp and on the propagation speed of the wave. Assuming an observation time, which is equal to the chirp duration Tb, to be at least ten times higher than the flight time Tp, the beat frequency fbeat will be resolvable to an accuracy of 2/Tb. As the detected range R is directly proportional to the detected beat frequency fbeat and the chirp duration Tb, the range resolution δfb is proportional to the resolution δfb of the beat frequency, which is equivalent to the frequency resolution. As the frequency resolution is inversely proportional to the chirp duration Tb, the range resolution δR is only dependant on the chirp bandwidth Δf and on the speed of propagation c, as it is analytically described in the following equation:
Thus, the range resolution δR is limited to c/(2Δf). In order to overcome the range resolution limitation, two main families of spectral estimation techniques can be considered: autoregressive modelling and time-domain eigenvalue decomposition. While autoregressive modelling might be a good candidate for single component estimation with high accuracy, time-domain eigenvalue decomposition is the favoured choice for range resolution enhancement in most applications.
The signal processing unit or method shown in
The initial step is a pre-estimation step.
This determined at least one frequency band of interest BoI and/or at least one signal-to-noise ratio SNRBoI is then used in at least one subsequent time-domain estimating step for accurately estimating, based on the time-domain signal y(t) or y(kTs), the at least one frequency fi and corresponding amplitude A, of the at least one exponential. This improves performance. Examples of such use of the BoI and/or SNRBoI will be given in the following description.
The at least one frequency band of interest BoI can, for example, be determined by determining a start frequency fa on one side with respect to the corresponding peak frequency fpeak and a stop frequency fb on the other side with respect to corresponding peak frequency fpeak. Thus, the BoI is a spectral range between the start frequency fa and the stop frequency fb. In this case, in a first example each of the start frequency fa and the stop frequency fb can be determined at a predefined distance from the corresponding peak frequency fpeak. Alternatively, in a second example each of the start frequency fa and the stop frequency fb can be determined at a predefined power level threshold PTH. Any other suitable way of determining the start frequency fa and the stop frequency fb can also be used.
Referring to the first example, a number of range bins Nbins, being a range bin the metric used to measure in the range axis, can be defined, in which the band of interest should consist on, and the start frequency fa and stop frequency fb of the band of interest are defined symmetrically from one peak frequency fpeak:
Referring to the second example, a power threshold PTH can be defined. Scanning from the frequency position of the nth peak fpeak
The same procedure is applied in the positive direction to obtain fb, wherein fs is the sampling frequency:
The expression for the nth BoI can then be the following:
BoI
n
: f ∈(f≧fa,n)&(f≦fb,n).
Thus, the SNR for the nth BoI can be written as:
The signal-to-noise ratio SNRBoI can for example be determined as the ratio of the maximum power level Y(fpeak) in the corresponding frequency band of interest BoI and the mean value of the power spectrum outside the corresponding frequency band of interest BoI.
In particular, a number of peak frequencies fpeak n can be determined. In this case a corresponding frequency band of interest BoIn for each peak frequency fpeak n, can be determined. Then, in one example a corresponding signal-to-noise ratio SNRBoI n for each frequency band of interest BoIn can be determined. In an alternative example, at least part of the number of frequency bands of interest can be joined to a joint frequency band of interest, and a corresponding signal-to-noise ratio for the joint frequency band of interest can be determined. The start frequency of the joint band of interest can be the start frequency of first BoI (from low to high frequency) and the stop frequency can be the stop frequency of last BoI.
As can be seen in
Turning again to
For example, in case a layer is detected at a short distance from the radar system, the detected peaks are at low frequencies. Up-conversion can then be applied to the signal before filtering in order to relax the specifications (order requirements) of the filter. Just as an example, if all the detected bands are at a relative frequency (normalized with half of the Nyquist frequency) lower than 0.1, the signal is modulated with a signal with relative frequency 0.1.
The time-domain eigenvalue decomposition (super-resolution technique) will now be explained in more detail.
100481 In this stage, a time-domain eigenvalue-decomposition of the time-domain signal y(t), y(kTs) is performed for determining the at least one complex exponential having the at least one estimated frequency f1 and corresponding amplitude Ai.
The objective of time-domain eigenvalue decomposition methods is to estimate a (uniformly sampled) time-domain signal X (kTs), in the presence of noise n(kTs), with at least one, or a group of (non equally spaced) complex exponentials, as shown in the following equation:
The time-domain eigenvalue decomposition can be formulated in multiple ways. For example, a MUSIC (Multiple Signal Classification (MUSIC) based method, or a Matrix-Pencil (MP) based method can be used. In particular, the Matrix-Pencil based method can be a direct Matrix-Pencil method or an Estimation of Parameters via Rotational Invariant Techniques (ESPRIT), which is a variation of the direct Matrix-Pencil method.
The Matrix-Pencil based method is now described in more detail. Assuming from a vector with uniformly sampled elements, y(0, 1, . . . , N−1), a projection matrix Y with size (N−L+1)×(L+1) is created as shown in the following:
From matrix Y two matrices Y1 and Y2 can be created, shown in the following:
Matrices Y, Y1 and Y2 have a contribution of a signal and a noise subspace and the objective is to extract the signal subspace, X, from Y in order to obtain the complex exponential components that define the time-domain signal which is being processed. Singular value decomposition (SVD) is used to separate these sub spaces. The SVD decomposition equation for matrix Y is show in the following equation:
Y=U·Σ·V
H.
The matrix U is a unitary matrix and represents the left singular vectors of Y, V is an orthogonal matrix and represents the right singular vectors of Y and finally Σ is a diagonal matrix with the singular values o related to the singular vectors. This decomposition can be applied likewise to matrices Y1 and Y2. Subsequently matrix Y and/or Y1 and Y2 depending on the implementation are truncated (represented by the subscript T) using the M biggest singular values and vectors. This is analytically described in the following equation:
Y
T
=U
T·ΣET·VTH.
M is the number of singular values related to the signal subspace (also called model-order) and L-M the smaller singular values related to the noise subspace. The number M of singular values that define the signal subspace is decided, in most of the state of the art published material by defining a tolerance factor c which is applied to the biggest singular value. M is determined as the number of singular values bigger than the tolerance factor times the biggest singular value. This assumes that the singular values are ordered by decreasing magnitude as it is analytically described in the following equation:
Now the direct Matrix-Pencil method will be considered, for example as proposed by Sarkar in Sarkar et al. “Using the Matrix Pencil Method to Estimate the Parameters of a Sum of Complex Exponentials”, IEEE Antennas and Propagation Magazine, vol. 37, no. 1, pp. 48-55, February 1995, which is incorporated by reference herein. The truncation is applied to both Y1 and Y2 and complex exponentials are obtained solving the eigenvalues from the following equation:
For the ESPRIT variation additional SVDs are realized and applied to the U and V vectors, as shown in the following:
[U1,U2]=UU·ΣU·└VU
The ESPRIT variation is for example described in Hua et al., “On SVD for Estimating Generalized Eigenvalues of Singular Matrix Pencil in Noise”, IEEE Transactions on Signal Processing, vol. 39, no. 4, pp. 892-900, April 1991, which is incorporated by reference herein.
The solution is obtained by extracting the eigenvalues of the following equation:
V
U
HΣ1VV
In each of the embodiments shown in
The embodiment of
wherein σi is the i-th singular value, L is the number of singular values, and εM is a tolerance factor. In this way, by using the determined signal-to-noise ratio, for example the robustness in low SNR situations or conditions can be improved.
Further, as can be seen in
Then, in
wherein Y is the projection matrix, H is a Hankel operator, Rank is a reduced-rank operator, F is the Frobenius norm, and εRRHA is a convergence value selected based on the determined signal-to-noise ratio SNRBoI. In an example, the determined signal-to-noise ratio SNRBoI can be compared with a third threshold TH3. The convergence value εRRHA is then set to a predefined value, if the determined signal-to-noise ratio SNRBoI is equal to or above the third threshold TH3. The convergence value εRRHA is a value depending on the quotient between the determined signal-to-noise ratio SNRBoI and the third threshold TH3, if the determined signal-to-noise ratio SNRBoI is below the third threshold TH3. This is another example where the SNRBoI is used in a subsequent processing step. In this way, for example the robustness in low SNR situations or conditions can be improved.
The model-order selection algorithm and the reduced-rank-Hankel approximation will now be explained in some further detail. After generation of the projection matrix and application of singular value decomposition, the singular values are used to estimate the number of components which define the signal subspace. This estimation can be then formulated as a model-order selection problem. One consideration has to be taken into account here. In case the time-domain signal is real, two complex exponentials are estimated for every spectral component. Otherwise, if the time-domain signal is complex, a single complex exponential is estimated for every component.
There are two types of model order estimation methods, a non-parametric (or “a priori”) and a parametric (or “a posteriori”) estimation type. This is for example described in Nadler, “Nonparametric Detection of Signals by Information Theoretic Criteria: Performance Analysis and an Improved Estimator”, IEEE Transactions on Signal Processing, vol. 58, no. 5, pp. 2746-2756, May 2010, which is incorporated by reference herein.
With the parametric type, a complete solution of the problem is calculated for different model order possibilities and based on the results the order of the problem is defined. With the non-parametric type, the order of the problem is estimated in an intermediate step. The parametric method is more accurate but requires higher complexity. The advantage of the non-parametric method is that lower complexity is required, but the resulting accuracy is worse. In order to maintain a low level of complexity, it is thus preferable to use the non-parametric model-order estimation method.
The non-parametric model order estimation uses the singular values (σ1, σ2, . . . , σL) of the leading diagonal of the diagonal matrix Σ, which was obtained during the SVD step:
Any suitable non-parametric method can be used. For example, an Information Theoretic Criterion (ITC) based method can be used. It provides a good solution when the SNR is high. In another example an Akaike Information Criterion (AIC), Minimum Description Length (MDL) or Efficient Description Criterion (EDC) based method can be used.
In particular, in the signal processing unit or method described herein, an EDC based method can be used, when the determined SNR is higher than TH2. In EDC a value is calculated for every possible model size to generate a vector:
In the first term of the equation, the natural logarithm is applied to the division between the geometric and the arithmetic mean of the singular values with higher index than the hypothetic model size M. The result of the logarithm is multiplied with the product between the size of the measured signal N and the difference between the size of the number of singular values obtained L and the hypothetic model size M. The second term of the equation is a penalty element which penalizes higher order estimations in front of lower order estimations. The second term depends on M, L and N. Once the vector is calculated for every possible model size, the index of the minimum value is extracted and this index is considered as the model-order of the problem.
An EDC based method is for example explained in R. Badeau, “A new perturbation analysis for signal enumeration in rotational invariance techniques”, IEEE Transactions on Signal Processing, vol. 54, no. 2, pp. 450-458, February 2006, which is incorporated by reference herein.
With low SNR environments, another method called gap criterion can show better results. Gap criterion is a method which uses the geometrical distance, also called gap, between the singular values in order to define the border between signal subspace and noise subspace. The basic idea is that the singular values related to the noise subspace are relatively similar between them, and the biggest gap between singular values exists between the signal and noise subspace. A gap criterion method is for example described in Liavas et al., “Blind Channel Approximation: Effective Channel Order Determination”, IEEE Transactions on Signal Processing, vol. 47, no. 12, pp. 3336-3344, December 1999, which is incorporated by reference herein.
By taking our estimate model size (or rank estimate for the signal subspace) to be {tilde over (M)}, the signal space can be represented by the matrix SM, and the noise sub-space can be represented by NM, as shown in the following equations, where r is a vector containing the gap distances between the singular eigenvalues:
In particular, in the signal processing unit or method describe herein, a variation of the gap criterion described above can be used, when the determined SNR is lower than TH2. In this variation, the following formulas can be used:
r′
i
=r
i+1
−r
i
M
A=arg maxi r′
r′
M
≦r′
M
·εM {tilde over (M)}=MB−1
The difference between consecutive elements ri (where i =1 , . . . , L) of the gap vector r is calculated. The index of the maximum position MA is extracted and a tolerance factor εM is applied to the element of r′ in the maximum position r′M
The rank truncation operator with truncation order M is:
It does not preserve the Hankel structure that a projection matrix has. An example of a Hankel matrix is:
For a Hankel matrix an element xi, j at row i and column j can be defined as xi,j=xi−1, j+1.
A consequence of ignoring this property is a degradation of the separation between signal and noise when the SNR conditions are low. The application of a single Hankel operator after a rank approximation is:
This does not solve the issue because, in this case, the reduced-rank property is not kept anymore, which means that no analytic operator can keep both properties at the same time. A solution for this issue is the application of the Reduced Rank Hankel Approximation (RRHA) algorithm to generate a matrix that approximates both properties as indicated in the following equation, where J stands for the RRHA operation:
Theoretically the algorithm only converges after infinite iterations, as shown in the above equation. Therefore this mathematical expression can not be implemented but approximated. In the signal processing unit and method described herein, the number of iterations of the RRHA algorithm is limited taking into account the pre-detected SNR and the difference between two consecutive iterations.
The convergence of the RRHA algorithm can be decided by analyzing the variation between two consecutive iterations. It is considered that the algorithm has converged, if the Frobenius norm between two consecutive iterations does not exceed a certain relative value c, also called convergence factor. The convergence of the RRHA algorithm can be by the following equation:
In the signal processing unit and method disclosed herein, the selection of E can be done taking into account the pre-detected SNR. If the detected SNR is TH3 (dB) or bigger, ε is set to a certain value, for example 1%. If the SNR is below TH3, the difference between TH3 and the actual SNR is used as a division factor to obtain the convergence factor ε. Just as an example, if the detected SNR is 20 dB and TH3 is 30 dB, the convergence factor ε is:
εactual=εTH3·10−(30−20)=0.1%, wherein εTH
After the application of RRHA, the solution is obtained solving the eigen-values as previously explained.
Returning to
The iterative optimization algorithm can for example be a (iterative) least squares algorithm with the at least one complex exponential, in particular multiple complex exponentials, as seeds. This iterative optimization algorithm may use the cost function:
f
cos t
=w
1·MSE(ryy,ryz)+w2·ryz(0),
wherein w1 and w2 is each a weight, MSE is a mean square error, ryy is an autocorrelation of the time-domain signal y(kTs), and ryx is a cross-correlation between the time-domain signal y(kTs) and the complex exponential signal. This cost function used in the optimization problem measures the difference between the (uniformly) sampled input vector, y, and the iteratively corrected synthesized signal with complex exponentials {tilde over (x)} ({tilde over (x)} being an estimation of x). As indicated in the above formula, first the mean square error between the autocorrelation ryy of the normalized input signal and the cross-correlation ryx between the normalized input and the normalized synthesized signal is calculated. Then, the cost function f cost is formed by weighting the mean square error MSE by the weight w1 and adding the result to the cross-correlation of y and x at zero offset weighted by the weight w2.
A stopping criterion of the iterative optimization algorithm may be at least one criterion selected from the group comprising reaching a predefined maximum number of iterations maxiter, determining that a variation of the estimated frequency between successive iterations is less than a frequency resolution divided by a predefined factor, and determining that a variation of the cost function fcost is below a minimum allowed variation of the cost function εcost. In particular, all three criteria can be used.
In case of the object detection system as explained with reference to
The minimum allowed variation of the cost function εcostcan depend on the determined signal-to-noise ratio SNRBoI. In particular, the minimum allowed variation of the cost function εcost can be a predefined value εcostTh
Just as an example, the stopping criterion can be the following three. The first stopping criterion can be determining that the variation of the range position in all the estimated exponentials from one iteration to the next one is smaller than the range resolution (applying frequency-domain techniques) c/(2Δf) divided with a predefined factor Rfact (e.g. 100 or the like). The second stopping criterion can be determining that the relative variation of the cost function εcost is below a minimum allowed variation of the cost function. This can for example be defined depending on the SNR. If the SNR is higher than TH5, εcostTH
The performance of the (time-domain super-resolution) algorithm described herein will now be explained with reference to
Obviously, numerous modifications and variations of the present disclosure are possible in light of the above teachings. It is therefore to be understood that within the scope of the appended claims, the invention may be practiced otherwise than as specifically described herein.
In the claims, the word “comprising” does not exclude other elements or steps, and the indefinite article “a” or “an” does not exclude a plurality. A single element or other unit may fulfill the functions of several items recited in the claims. The mere fact that certain measures are recited in mutually different dependent claims does not indicate that a combination of these measures cannot be used to advantage.
In so far as embodiments of the invention have been described as being implemented, at least in part, by software-controlled data processing apparatus, it will be appreciated that a non-transitory machine-readable medium carrying such software, such as an optical disk, a magnetic disk, semiconductor memory or the like, is also considered to represent an embodiment of the present invention. Further, such a software may also be distributed in other forms, such as via the Internet or other wired or wireless telecommunication systems.
It follows a list of further embodiments:
1. A signal processing unit for accurately estimating from a time-domain signal (y(t), y(kTs)) at least one frequency (fi) and corresponding amplitude (Ai) of at least one complex exponential, the signal processing unit configured to:
wherein σi is the i-th singular value, L is the number of singular values, and εM is a tolerance factor.
16. The signal processing unit of one of embodiments 9 to 15, further configured to perform a reduced-rank-Hankel-approximation (RRHA) algorithm.
17. The signal processing unit of embodiment 16, wherein the estimated model order (M) is used in the reduced-rank-Hankel approximation (RRHA) algorithm.
18. The signal processing unit of embodiment 16 or 17, wherein the determined signal-to-noise ratio (SNRBoI) is used in the reduced-rank-Hankel approximation (RRHA) algorithm.
19. The signal processing unit of one of embodiments 16 to 18, wherein the stopping criterion of the reduced-rank-Hankel approximation (RHHA) algorithm is
wherein Y is the projection matrix, H is a Hankel operator, Rank is a reduced-rank operator, F is the Frobenius norm, and εRRHA is a convergence value selected based on the determined signal-to-noise ratio (SNRBoI).
20. The signal processing unit of embodiment 19, configured to compare the determined signal-to-noise ratio (SNRBoI) with a third threshold (TH3), wherein the convergence value (εRRHA) is set to a predefined value, if the determined signal-to-noise ratio (SNRBoI) is equal to or above the third threshold (TH3), and wherein the convergence value (εRRHA) is a value depending on the quotient between the determined signal-to-noise ratio (SNRBoI) and the third threshold (TH3), if the determined signal-to-noise ratio (SNRBoI) is below the third threshold (TH3).
21. The signal processing unit of one of embodiments 9 to 20, configured to refine the at least one estimated frequency of the at least one complex exponential signal by performing an iterative optimization algorithm.
22. The signal processing unit of embodiment 21, configured to compare the signal-to-noise ratio (SNRBoI) with a fourth threshold (TH4), and to perform the iterative optimization algorithm, only if the signal-to-noise ratio (SNRBoI) is equal to or above the fourth threshold (TH4).
23. The signal processing unit of embodiment 21 or 22, wherein the iterative optimization algorithm is a least squares algorithm with the at least one complex exponential as seeds.
24. The signal processing unit of one of embodiments 21 to 23, wherein the iterative optimization algorithm uses the cost function
f
cost
=w
1·MSE(ryy,ryx)+w2·ryx(0),
wherein w1 and w2 is each a weight, MSE is a mean square error, ryy is an autocorrelation of the time-domain signal (y(kTs)), and ryx is a cross-correlation between the time-domain signal (y(kTs)) and the complex exponential signal.
25. The signal processing unit of one of embodiments 21 to 24, wherein a stopping criterion of the iterative optimization algorithm is at least one criterion selected from the group comprising reaching a predefined maximum number of iterations (maxiter), determining that a variation of the estimated frequency between successive iterations is less than a frequency resolution divided by a predefined factor, and determining that a variation of the cost function (fcost) is below a minimum allowed variation of the cost function (εcost).
26. The signal processing unit of embodiments 25, wherein the minimum allowed variation of the cost function (εcost) depends on the determined signal-to-noise ratio (SNRBoI).
27. The signal processing unit of embodiment 26, wherein the minimum allowed variation of the cost function (εcost) is a predefined value (εcostTH
28. The signal processing unit of one of the preceding embodiments, configured to detect a number of peak frequencies (fpeak n), to determine a corresponding frequency band of interest (BoIn) for each peak frequency (fpeak n), and to determine a corresponding signal-to-noise ratio (SNRBoI n) for each frequency band of interest (BoIn).
29. The signal processing unit of one of embodiments 1 to 27, configured to detect a number of peak frequencies (fpeak n), to determine a corresponding frequency band of interest (BoIn) for each peak frequency (fpeak n), to join at least part of the number of frequency bands of interest to a joint frequency band of interest, and to determine a corresponding signal-to-noise ratio for the joint frequency band of interest.
30. An object detection system (10) for detecting at least one target object (1) at a range (R), the system comprising:
Number | Date | Country | Kind |
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11169553.2 | Jun 2011 | EP | regional |