# Patent

## Title
Delay restricted channel estimation for multi-carrier systems

## Abstract

A method is disclosed that includes channel estimation for a multi-carrier modulation of a plurality of subcarriers. The method includes performing a first channel estimation to obtain a first estimate of a channel based on a received signal. The method further includes suppressing a portion of elements of the first estimate of the channel to obtain a suppressed estimate of the channel. The method also includes multiplying the suppressed estimate of the channel by a first matrix to obtain a delay restricted estimate of the channel.

## Background
PRIORITY
The present application claims priority from and is a continuation of patent application Ser. No. 11/046,192, filed on January 28, now U.S. Pat No. 7,551,547, and entitled “DELAY RESTRICTED CHANNEL ESTIMATION FOR MULTI-CARRIER SYSTEMS,” the contents of which are expressly incorporated herein by reference in its entirety.
FIELD OF THE DISCLOSURE
The present disclosure relates to methods and systems for channel estimation of a wireless communication channel.
BACKGROUND
In recent years, Orthogonal Frequency Division Multiplexing (OFDM) has attracted attention as a candidate for high-data-rate video and multimedia communications. OFDM belongs to a family of transmission schemes called multi-carrier modulation. Multi-carrier modulation is based on dividing a given high bit-rate data stream into several parallel low bit-rate data streams and modulating each stream on separate sub-carriers.
The motivation for using multi-carrier modulation is to overcome the problem of inter-symbol interference (ISI). In wireless channels where radio signals from a transmitter traverse multiple paths to a receiver, all the signal energy does not necessarily arrive at the receiver at the same instant of time. This phenomenon of dispersion in a communication channel causes energy from one symbol duration to spill into succeeding symbol durations.
When a time delay due to dispersion is either a significant fraction of or more than the symbol time duration, the resultant ISI can be detrimental. ISI causes an irreducible error floor that cannot be overcome by simply changing a radio frequency (RF) parameter such as a transmit power, an antenna pattern or a frequency plan.
In an OFDM system, each sub-carrier can be viewed as a flat fading channel. A single tap equalizer can be used to equalize the transmitted signal in case of coherent demodulation. This requires the receiver to have knowledge of the channel on a per-sub-carrier basis.
The discrete baseband time representation of a transmitted OFDM signal is:




  
    
      
        
          y
          
            n
            ,
            m
          
        
        =
        
          
            ∑
            
              k
              =
              o
            
            
              N
              -
              1
            
          
          ⁢
          
            
              x
              
                k
                ,
                m
              
            
            ⁢
            
              e
              
                j
                ⁢
                
                  
                
                ⁢
                2
                ⁢
                
                  
                
                ⁢
                
                  π
                  ⁡
                  
                    (
                    
                      k
                      -
                      
                        N
                        2
                      
                    
                    )
                  
                
                ⁢
                
                  n
                  N
                
              
            
          
        
      
    
    
      
        (
        1
        )
      
    
  




where xk,m represents the sub-carrier symbols, N is the number of sub-carriers, k is the modulated symbol index, m is the OFDM symbol index, n is a sub-carrier index, and j represents the square root of −1.

The discrete version of the OFDM baseband signal shown in equation (1) is identical to the Inverse Discrete Fourier Transform (IDFT) of the sub-carrier symbols xn,m. Thus, OFDM modulation is essentially identical to an IDFT operation which may be performed using an Inverse Fast Fourier Transform (IFFT). FIG. 1 (prior art) is a block diagram of a baseband representation of an OFDM transmitter. The OFDM transmitter comprises a serial-to-parallel converter 10, an IFFT processor 12, and a parallel-to-serial converter 14.
FIG. 2 (prior art) is a graph illustrating frequency synchronization of an OFDM signal. The graph illustrates spectral component versus frequency for the case of using four tones.
The sub-carrier data symbols can be estimated at a receiver by taking the Discrete Fourier Transform (DFT) such as a Fast Fourier Transform (FFT) of a received and equalized OFDM symbol using the following equation.




  
    
      
        
          
            x
            ^
          
          
            n
            ,
            m
          
        
        =
        
          
            1
            N
          
          ⁢
          
            
              ∑
              
                k
                =
                o
              
              
                N
                -
                1
              
            
            ⁢
            
              
                y
                
                  k
                  ,
                  m
                
              
              ⁢
              
                e
                
                  
                    -
                    j
                  
                  ⁢
                  
                    
                  
                  ⁢
                  2
                  ⁢
                  
                    
                  
                  ⁢
                  
                    π
                    ⁡
                    
                      (
                      
                        n
                        -
                        
                          N
                          2
                        
                      
                      )
                    
                  
                  ⁢
                  
                    k
                    N
                  
                
              
            
          
        
      
    
    
      
        (
        2
        )
      
    
  




In a time dispersive channel such as one which introduces multipath fading, the signal at the receiver can be written as a convolution of the transmitted signal y and the channel impulse response h. Thus, the received signal rn,m in the discrete time domain can be written as:




  
    
      
        
          r
          
            n
            ,
            m
          
        
        =
        
          
            
              ∑
              
                l
                =
                o
              
              
                L
                -
                1
              
            
            ⁢
            
              
                h
                l
              
              ⁢
              
                y
                
                  
                    n
                    -
                    l
                  
                  ,
                  m
                
              
              ⁢
              
                
              
              ⁢
              n
            
          
          ≥
          L
        
      
    
    
      
        (
        
          3
          ⁢
          a
        
        )
      
    
  
  
    
      
        
          r
          
            n
            ,
            m
          
        
        =
        
          
            
              
                ∑
                
                  l
                  =
                  o
                
                n
              
              ⁢
              
                
                  h
                  l
                
                ⁢
                
                  y
                  
                    
                      n
                      -
                      l
                    
                    ,
                    m
                  
                
              
            
            +
            
              
                ∑
                
                  l
                  =
                  
                    n
                    +
                    1
                  
                
                
                  L
                  -
                  1
                
              
              ⁢
              
                A
                ⁢
                
                  
                
                ⁢
                
                  h
                  l
                
                ⁢
                
                  y
                  
                    
                      N
                      -
                      
                        (
                        
                          l
                          -
                          n
                        
                        )
                      
                    
                    ,
                    
                      m
                      -
                      l
                    
                  
                
                ⁢
                
                  
                
                ⁢
                n
              
            
          
          <
          L
        
      
    
    
      
        (
        
          3
          ⁢
          b
        
        )
      
    
  




where L is the length of the channel impulse response in units of sample time.

No ISI is observed in received signal when n=L. However when n<L, the signal from the previous OFDM symbol is corrupted by the received signal. If the cyclic prefix is longer than the channel impulse response, then the effect of the previous OFDM symbol is not seen in the usable part of the signal (i.e. the part of the OFDM symbol after the cyclic prefix). From here onward, it is assumed herein that the cyclic prefix is longer than the channel impulse response, thus ignoring the second term in equation (3b). Under this assumption, the demodulated signal in each sub-carrier is given by:




  
    
      
        
          
            
              
                s
                
                  n
                  ,
                  m
                
              
              =
              
              ⁢
              
                
                  1
                  N
                
                ⁢
                
                  
                    ∑
                    
                      k
                      =
                      0
                    
                    
                      N
                      -
                      1
                    
                  
                  ⁢
                  
                    
                      r
                      
                        k
                        ,
                        m
                      
                    
                    ⁢
                    
                      e
                      
                        
                          -
                          j
                        
                        ⁢
                        
                          
                        
                        ⁢
                        2
                        ⁢
                        
                          
                        
                        ⁢
                        
                          π
                          ⁡
                          
                            (
                            
                              n
                              -
                              
                                N
                                2
                              
                            
                            )
                          
                        
                        ⁢
                        
                          k
                          N
                        
                      
                    
                  
                
              
            
          
        
        
          
            
              =
              
              ⁢
              
                
                  1
                  N
                
                ⁢
                
                  
                    ∑
                    
                      k
                      =
                      0
                    
                    
                      N
                      -
                      1
                    
                  
                  ⁢
                  
                    
                      ∑
                      
                        l
                        =
                        0
                      
                      
                        L
                        -
                        1
                      
                    
                    ⁢
                    
                      
                        h
                        l
                      
                      ⁢
                      
                        y
                        
                          
                            k
                            -
                            l
                          
                          ,
                          m
                        
                      
                      ⁢
                      
                        e
                        
                          
                            -
                            j
                          
                          ⁢
                          
                            
                          
                          ⁢
                          2
                          ⁢
                          
                            
                          
                          ⁢
                          
                            π
                            ⁡
                            
                              (
                              
                                n
                                -
                                
                                  N
                                  2
                                
                              
                              )
                            
                          
                          ⁢
                          
                            k
                            N
                          
                        
                      
                    
                  
                
              
            
          
        
        
          
            
              =
              
              ⁢
              
                
                  λ
                  n
                
                ⁢
                
                  x
                  
                    n
                    ,
                    m
                  
                
              
            
          
        
      
    
    
      
        (
        4
        )
      
    
  




where λn is the DFT of the channel impulse response.

Even though the channel is time dispersive, the effect of the channel can be visualized in the frequency domain (i.e. after the DFT) by a single multiplicative constant λn for each of the sub-carriers. This results because a convolution operation in the time domain translates to a simple multiplication in the frequency domain. A single-tap equalization using a zero-forcing equalizer can be used to estimate the sub-carrier signal using the following equation.




  
    
      
        
          
            x
            ^
          
          
            n
            ,
            m
          
        
        =
        
          
            s
            
              n
              ,
              m
            
          
          
            λ
            
              n
              ,
              m
            
          
        
      
    
    
      
        (
        5
        )
      
    
  




However, in order to use equation (5), the receiver needs to know the channel response in the frequency domain (i.e. the values of λn).
The channel response may be estimated either by embedding pilot symbols within data stream or by using preambles. The known pilot or preamble symbols are used to estimate the channel on the given sub-carriers. To estimate the channel in other sub-carriers, a subsequent channel interpolation can be performed.
Existing channel estimation techniques include zero forcing and linear minimum mean square error (LMMSE). Zero-forcing channel estimation can be performed over known pilot and/or preamble symbols by dividing the received symbol by the expected symbol.




  
    
      
        
          
            λ
            ^
          
          n
        
        =
        
          
            s
            
              n
              ,
              m
            
          
          
            x
            
              n
              ,
              m
            
          
        
      
    
    
      
        (
        6
        )
      
    
  




A disadvantage of a zero forcing estimate is its unreliability in low signal-to-noise ratio (SNR) conditions. For example, a zero-forcing-estimated channel response can be significantly inaccurate if some of the sub-carriers experience a deep fade.
The LMMSE channel estimator is designed to minimize the mean square error between an estimated channel response and an actual channel response. For convenience in formulating the LMMSE estimator, the relationship between received and transmitted symbols in the sub-carriers carrying the pilot/preamble is represented in the following vector form:




  
    
      
        
          S
          =
          
            
              P
              ⁢
              
                
              
              ⁢
              Λ
            
            +
            W
          
        
        ,
        
          
            or
            ⁢
            
              

            
            [
            
              
                
                  
                    s
                    0
                  
                
              
              
                
                  
                    s
                    1
                  
                
              
              
                
                  ⋮
                
              
              
                
                  
                    s
                    
                      M
                      -
                      1
                    
                  
                
              
            
            ]
          
          =
          
            
              
                [
                
                  
                    
                      
                        p
                        0
                      
                    
                    
                      0
                    
                    
                      ...
                    
                    
                      0
                    
                  
                  
                    
                      0
                    
                    
                      
                        p
                        1
                      
                    
                    
                      
                        
                      
                    
                    
                      0
                    
                  
                  
                    
                      ⋮
                    
                    
                      
                        
                      
                    
                    
                      ⋱
                    
                    
                      ⋮
                    
                  
                  
                    
                      0
                    
                    
                      0
                    
                    
                      ...
                    
                    
                      
                        p
                        
                          M
                          -
                          1
                        
                      
                    
                  
                
                ]
              
              ⁡
              
                [
                
                  
                    
                      
                        λ
                        0
                      
                    
                  
                  
                    
                      
                        λ
                        1
                      
                    
                  
                  
                    
                      ⋮
                    
                  
                  
                    
                      
                        λ
                        
                          M
                          -
                          1
                        
                      
                    
                  
                
                ]
              
            
            +
            
              [
              
                
                  
                    
                      w
                      0
                    
                  
                
                
                  
                    
                      w
                      1
                    
                  
                
                
                  
                    ⋮
                  
                
                
                  
                    
                      w
                      
                        M
                        -
                        1
                      
                    
                  
                
              
              ]
            
          
        
      
    
    
      
        (
        7
        )
      
    
  




where si and pi are the received and transmitted symbols in the sub-carriers carrying the pilot/preamble, and wi represents noise. The noise can be assumed to be additive, Gaussian white noise.

The LMMSE estimate {circumflex over (Λ)} of the channel impulse response is determined by the following equation:

{circumflex over (Λ)}=AS=AP{circumflex over (Λ)}+AW  (8)

where A is an estimation matrix.

The estimation matrix A is determined by the following equation:

A=Rλ[Rλ+(PHP)−1Σ]−1P−1  (9)

where Rλ is the covariance matrix of the channel impulse response and Σ is the covariance matrix of the noise vector and usually is a diagonal matrix.

In the absence of noise, the covariance matrix Σ is equal to a zero matrix. If the fading in the different sub-carriers is independent, the covariance matrix Rλ of the channel impulse response is an identity matrix. Under both of these two conditions, the LMMSE estimate is identical to the zero forcing estimate.
A drawback of the LMMSE channel estimation scheme is its requiring knowledge of the correlation between the fades of different sub-carriers to form the covariance matrix Rλ. In most practical systems, this information is not known at the receiver a priori, thus making the LMMSE estimator impractical.
Channel estimation and equalization are fundamental components of wireless communication systems, especially those that have been designed to work in a non-line-of-sight condition. In multi-carrier systems such as OFDM, equalization is relatively straightforward, but sophisticated channel estimation and channel interpolation techniques are presently required.



BRIEF DESCRIPTION OF THE DRAWINGS
The present invention is pointed out with particularity in the appended claims. However, other features are described in the following detailed description in conjunction with the accompanying drawings in which:
FIG. 1 (prior art) is a block diagram of a baseband representation of an OFDM transmitter;
FIG. 2 (prior art) is a graph illustrating frequency synchronization of an OFDM signal;
FIG. 3 is a flow chart summarizing an embodiment of a method of delay-restricted channel estimation;
FIGS. 4 and 5 show symbol/sub-carrier combinations in pilot and preambles that can be used for delay-restricted channel estimation;
FIG. 6 is a block diagram of an embodiment of a communication system having a delay-restricted channel estimator; and
FIG. 7 illustrates graphs of throughput versus signal-to-noise ratio for simulations of the delay restricted channel estimation method.



DETAILED DESCRIPTION
A method is disclosed that includes channel estimation and includes performing a first channel estimation to obtain a first estimate of a channel based on a received signal. The method further includes suppressing a portion of elements of the first estimate of the channel to obtain a suppressed estimate of the channel. The method also includes multiplying the suppressed estimate of the channel by a first matrix to obtain a delay restricted estimate of the channel.
In another particular embodiment, a receiver includes a first channel estimator to compute a first channel estimation based on a received signal. The receiver also includes a suppressor to suppress a portion of elements of the first channel estimation to obtain a suppressed channel estimation of the channel. The receiver further includes a multiplier to multiply the suppressed channel estimation by a first matrix to obtain a delay restricted estimate of the channel.
In another particular embodiment, a computer readable storage medium comprises operational instructions that, when executed by a processor, cause the processor to perform a first channel estimation to obtain a first estimate of a channel based on a received signal. The computer readable storage medium also includes operational instructions that, when executed by the processor, cause the processor to suppress a portion of elements of the first estimate of the channel to obtain a suppressed estimate of the channel. The computer readable storage medium also includes operational instructions that, when executed by the processor, cause the processor to multiply the suppressed estimate of the channel by a first matrix to obtain a delay restricted estimate of the channel.
Existing frequency domain channel estimation schemes for OFDM do not incorporate information that the channel impulse response exists only over a finite number of samples in the time domain because the delay spread of the environment is finite and much smaller than the OFDM symbol duration. This assumption can be made since the delay spread of most environments is of the order of a few microseconds, e.g. 15-20 microseconds in some environments. Thus, in the time domain, a span of channel impulse response is essentially limited.
Described herein are embodiments of delay restricted channel estimation methods and systems that take into account a finite spread of the channel impulse response in the time domain to improve the accuracy of channel estimation in the frequency domain. The new methods can be used for preamble-based or pilot-based channel estimation.
The disclosed methods provide channel estimates that are less susceptible to noise. As a benefit, the disclosed methods also allow for estimation of a multi-carrier noise variance estimation. Simulation results show a 2 dB improvement in the link performance by implementing a particular disclosed channel estimation method when compared to traditional channel estimation schemes such as zero forcing and LMMSE.
The delay restricted channel estimation can be based on a zero-forcing estimate, an LMMSE estimate, or an alternative estimate with an additional constraint that the impulse response of the channel exists only over a finite and known interval of time.
A vector Λ comprising the values λn from equation (4) is related to a DFT matrix F and a channel impulse response vector H by the following equation:

Λ=FH, or





  
    
      
        
          [
          
            
              
                
                  λ
                  0
                
              
            
            
              
                
                  λ
                  1
                
              
            
            
              
                ⋮
              
            
            
              
                
                  λ
                  
                    N
                    -
                    1
                  
                
              
            
          
          ]
        
        =
        
          
            [
            
              
                
                  
                    F
                    
                      0
                      ,
                      0
                    
                  
                
                
                  ...
                
                
                  
                    F
                    
                      0
                      ,
                      
                        L
                        -
                        1
                      
                    
                  
                
              
              
                
                  ⋮
                
                
                  ⋱
                
                
                  ⋮
                
              
              
                
                  
                    F
                    
                      
                        N
                        -
                        1
                      
                      ,
                      0
                    
                  
                
                
                  ...
                
                
                  
                    F
                    
                      
                        N
                        -
                        1
                      
                      ,
                      
                        L
                        -
                        1
                      
                    
                  
                
              
            
            ]
          
          ⁡
          
            [
            
              
                
                  
                    h
                    0
                  
                
              
              
                
                  ⋮
                
              
              
                
                  
                    h
                    
                      L
                      -
                      1
                    
                  
                
              
            
            ]
          
        
      
    
    
      
        (
        10
        )
      
    
  




The elements of the DFT matrix F are given by:




  
    
      
        
          
            F
            
              i
              ,
              k
            
          
          =
          
            e
            
              
                -
                j
              
              ⁢
              
                
              
              ⁢
              2
              ⁢
              
                
              
              ⁢
              
                π
                ⁡
                
                  (
                  
                    i
                    -
                    
                      N
                      2
                    
                  
                  )
                
              
              ⁢
              
                k
                N
              
            
          
        
        ,
        
          
        
        ⁢
        
          
            for
            ⁢
            
              
            
            ⁢
            k
          
          <
          L
        
      
    
    
      
        (
        11
        )
      
    
  




The DFT matrix F is an N×L matrix of rank L, where L is the number of taps over which the channel impulse response exists. A singular value decomposition of the DFT matrix F can be performed to determine a unitary transformation U and V such that

F=U FVH F=UHFV  (12)

where F is an N×L diagonal matrix containing the singular values of F, and U and V are N×N and L×L dimensional matrices respectively with unitary columns.

The matrices U and V can be viewed as unitary transformation operators in the frequency and time domains, respectively. In particular, the vector Λ can be pre-multiplied by UH (i.e. the complex conjugate transpose of U) to form a transformed vector Λ. The vector H can be pre-multiplied by VH (i.e. the complex conjugate transpose of V) to form a transformed vector H. The relationship between Λ, F and H is represented by the following equation:

 Λ= FH  (13)

The matrix F has the following form.




  
    
      
        
          F
          _
        
        =
        
          [
          
            
              
                
                  
                    F
                    _
                  
                  0
                
              
              
                ...
              
              
                0
              
            
            
              
                ⋮
              
              
                
                  
                
              
              
                ⋮
              
            
            
              
                0
              
              
                ...
              
              
                
                  
                    F
                    _
                  
                  
                    L
                    -
                    1
                  
                
              
            
            
              
                0
              
              
                ...
              
              
                0
              
            
            
              
                ⋮
              
              
                
                  
                
              
              
                ⋮
              
            
            
              
                0
              
              
                ...
              
              
                0
              
            
          
          ]
        
      
    
    
      
        (
        14
        )
      
    
  




Based on the form of F, the elements after the first L elements in the vector Λ are all zero.




  
    
      
        
          Λ
          _
        
        =
        
          [
          
            
              
                
                  
                    λ
                    _
                  
                  0
                
              
            
            
              
                ⋮
              
            
            
              
                
                  
                    λ
                    _
                  
                  
                    L
                    -
                    1
                  
                
              
            
            
              
                0
              
            
            
              
                ⋮
              
            
            
              
                0
              
            
          
          ]
        
      
    
    
      
        (
        15
        )
      
    
  




The unitary transformation U is used to modify a channel estimate, such as either the zero forcing or the LMMSE channel estimate, as follows. Once the estimate Λ has been evaluated, the elements λi for i≧L are suppressed to zero based on the form of Λ given by equation (15). The transformation U is used to transform Λ back to Λ in the frequency domain, i.e. U is multiplied by Λ to obtain a delay-restricted estimate of Λ.
To show the derivation, both sides of equation (7) are multiplied by UH to transform the received symbol vector S, the transmitted symbol matrix P, the Λ vector and the noise vector W.

UHS=UHPUUHΛ+UHW  (16)

Thus, equation (16) relates a transformed received symbol vector S, a transformed transmitted symbol matrix P, the transformed vector Λ, and the transformed noise vector W according to the following equation.

 S= P Λ+ W  (17)

Since the transformation is unitary, the covariance matrices of the noise vector W and the transformed noise vector W are identical.
The delay-restricted zero-forcing channel estimate can be determined by: (i) determining the transformed vector Λ based on the equation:

 Λ= P−1 S  (18)

(ii) suppressing all but the first L elements in Λ to zero; and (iii) transforming Λ back to Λ using the following equation:

Λ=U Λ  (19)

Similarly the delay-restricted LMMSE channel estimate can be determined by: (i) determining the transformed vector Λ based on the equation:

 Λ=R λ[R λ+( PH P)−1Σ]−1 P−1 S  (20)

(ii) suppressing all but the first L elements in Λ to zero; and (iii) transforming Λ back to Λ using the following equation:

Λ=U Λ  (21)

A traditional zero-forcing channel estimate does not allow noise variation estimation to be performed. However, by taking advantage of the finite spread of the channel impulse response in the time domain, the delay-restricted zero-forcing channel estimate enables estimation of a noise variance. The estimate of noise W is determined by subtracting the estimated signal from the received signal.

 W=r−P Λ  (22)

A noise variance σ2 is determined by an inner product of WH and W.

σ2= WH W  (23)

This approach to noise variance estimation is unsuitable for the case of a traditional zero-forcing estimate since a zero noise variance will result based on the definition of the zero-forcing estimate.
FIG. 3 is a flow chart summarizing an embodiment of a method of delay restricted channel estimation. As indicated by block 20, the method comprises determining the transformation matrix UH. The transformation matrix UH can be determined by performing a singular value decomposition of the DFT matrix, F, to determine the matrix U, and then determining the complex conjugate transpose of U.
As indicated by block 22, the method comprises transforming the transmitted symbol matrix P based on the transformation matrix UH. The transformed transmitted symbol matrix P is equal to UH multiplied by P.
As indicated by block 24, the method comprises transforming the received symbol vector S based on the transformation matrix UH. The transformed received symbol vector S is equal to UH multiplied by S.
As indicated by block 26, the method comprises determining a transformed vector Λ based on the transformed transmitted symbol matrix P and the transformed received symbol vector S. The transformed vector Λ is determined using a channel estimation method such as zero-forcing, LMMSE, or an alternative method. For zero-forcing, Λ is determined using equation (18); for LMMSE, Λ is determined using equation (20).
As indicated by block 30, the method comprises suppressing all but the first L elements of the transformed vector Λ to zero. Thus, the first L elements (i.e. elements 1, 2, . . . , L) of the transformed vector Λ remain the same, and the remaining elements (i.e. elements L+1, L+2, . . . , N) are set to zero. It is noted that the scope of this disclosure includes variations wherein all elements after the Lth element are not suppressed exactly to zero, but substantially to zero.
As indicated by block 32, the method comprises inverse transforming the suppressed, transformed vector Λ based on the matrix U (which is an inverse transformation of the matrix UH) to determine a delay-restricted estimate of Λ. The delay-restricted estimate of Λ is equal to U multiplied by Λ (see equations (19) and (21)). Although the (L+1)st to the Nth elements in the transformed vector Λ are zero, the corresponding elements in Λ are not necessarily zero. Moreover, it is likely that all N elements of Λ are non-zero.
The method can be used to estimate the channel impulse response based on either pilots or preambles having symbol/sub-carrier combinations that are known a priori. FIGS. 4 and 5 show examples of pilots and preambles where symbol/sub-carrier combinations that are known a priori are shaded.
FIG. 6 is a block diagram of an embodiment of a communication system having a delay-restricted channel estimator 40. A transmitter 42 modulates a signal using a multi-carrier modulation method such as OFDM. The modulated signal is transmitted over a communication channel 44 to a receiver 46. The receiver 46 comprises the delay-restricted channel estimator 40, which performs an embodiment of the method described with reference to FIG. 3 for known pilots and/or preambles contained in the transmitted signal. The delay-restricted channel estimator 40 outputs the Λ vector to an equalizer 50. The equalizer 50 equalizes the received signal based on the Λ vector. A demodulator 52 demodulates an equalized signal from the equalizer 50.
Simulations of the delay-restricted channel estimation method were performed for an IEEE 802.16(d) system. The IEEE 802.16(d) system is a 256 sub-carrier based OFDM system that has been designed to provide high data rate and high QoS for wireless broadband access.
A 3GPP-based multipath channel (TS 25.996) was used to simulate the effect of a wireless non-line-of-sight (non-LOS) channel on a baseband signal. At the receiver, realistic channel estimation and frequency synchronization algorithms were used to keep the simulation close to real world performance of such systems.
TABLE I shows the parameters of the simulations.








TABLE I







Channel Bandwidth
5 MHz


Oversampling
7/6


Modulation
BPSK, QPSK, 16 QAM, & 64 QAM


Coding
Concatenated RS and Convolution Coding


Wireless Channel
Frequency Selective Fading (based on 3 GPP MIMO



Channel)


Doppler Spread
2 Hz (1 km/hr @ 2100 MHz)


Delay Spread
12 μsec


Transmit Diversity
Yes (2 antennas)


Receive Diversity
Yes (2 antennas)


Channel Estimation
Delay restricted and zero-forcing channel estimation


Noise Variance
Realistic


Estimation


Symbol Decoding
Soft


Synchronization
Ideal


MAC PDU Length
Variable


MAC SDU Length
Fixed (49 or 52 bytes)


Carrier Frequency
2300 MHz


Bandwidth
5 MHz









FIG. 7 are graphs of throughput versus signal-to-noise ratio for simulations of the delay restricted channel estimation method. Graphs 60 and 62 illustrate the performance of delay-restricted and conventional zero-forcing channel estimation, respectively, for QPSK modulation. Graphs 64 and 66 illustrate the performance of delay-restricted and conventional zero-forcing channel estimation, respectively, for 16-QAM modulation. Graphs 70 and 72 illustrate the performance of delay-restricted and conventional zero-forcing channel estimation, respectively, for 64-QAM modulation.
The graphs show that the delay-restricted channel estimation has improved throughput compared to conventional zero-forcing channel estimation for QPSK, 16-QAM and 64-QAM modulation. The overall performance of the system improves by about 2 dB by using the delay restricted channel estimation algorithm, which implies that the system requires a signal to noise ratio (SNR) 2 dB lower than what would be required if this channel estimation algorithm was not used. A significant portion of the gain comes from more accurate channel estimation and significantly better noise variance estimation.
The herein-disclosed delay restricted channel estimation method and system can be implemented in a fixed wireless access system. In one embodiment, the fixed wireless access system is based on the WiMAX standard. This implementation would lead to an improvement in coverage and capacity in the fixed wireless access system.
The herein-disclosed delay-restricted channel estimation method and system can be implemented in mobile radio telephones (e.g. cellular telephones) and/or mobile radio telephone base stations. In one embodiment, the delay-restricted channel estimation method and system are implemented by one or more integrated circuits for mobile telephones (e.g. a mobile telephone chip set).
Generally, the acts and components described herein can be implemented using a computer system. The computer system is directed by computer program code stored by a computer-readable storage medium to perform the acts described.
It will be apparent to those skilled in the art that the disclosed embodiments may be modified in numerous ways and may assume many embodiments other than the particular forms specifically set out and described herein.
The above-disclosed subject matter is to be considered illustrative, and not restrictive, and the appended claims are intended to cover all such modifications, enhancements, and other embodiments that fall within the true scope of the present invention. Thus, to the maximum extent allowed by law, the scope of the present invention is to be determined by the broadest permissible interpretation of the following claims and their equivalents, and shall not be restricted or limited by the foregoing detailed description.

## Claims

1. A method of estimating a channel at a receiver, the method comprising:
performing, at a channel estimator of the receiver, a first channel estimation to obtain a first estimate of the channel based on a received signal;
suppressing a portion of elements of the first estimate of the channel to obtain a suppressed estimate of the channel; and
multiplying the suppressed estimate of the channel by a first matrix to obtain a delay restricted estimate of the channel,
wherein the first estimate of the channel is determined from an inverse of a pilot symbol diagonal matrix multiplied by a vector of received symbols multiplied by a conjugate transpose of the first matrix.

2. The method of claim 1, wherein the first matrix is unitary.

3. The method of claim 2, wherein the unitary matrix is obtained by performing a singular value decomposition of a Fourier transform matrix.

4. The method of claim 1, wherein the first matrix is obtained from an operation on a second matrix.

5. The method of claim 4, wherein the second matrix is a transform matrix to transform a set of symbols from a first domain representation to a second domain representation.

6. The method of claim 5, wherein the transform matrix is a Fourier transform matrix.

7. The method of claim 4, wherein the operation includes a singular value decomposition.

8. The method of claim 1, further comprising determining a set of symbols of the received signal based on the delay restricted estimate of the channel.

9. The method of claim 8, wherein determining the set of symbols comprises dividing each received symbol of the received signal by a particular element of the delay restricted channel estimate.

10. The method of claim 1, further comprising estimating a noise variance from a noise estimate obtained by subtracting an estimated signal from the received signal.

11. The method of claim 10, wherein the estimated signal is determined by multiplying a preamble symbol matrix by the delay restricted estimate of the channel.

12. The method of claim 1, wherein suppression of an element comprises setting the element to zero.

13. A receiver, comprising:
a first channel estimator to compute a first channel estimation based on a received signal;
a suppressor to suppress a portion of elements of the first channel estimation to obtain a suppressed channel estimation of the channel; and
a multiplier to multiply the suppressed channel estimation by a first matrix to obtain a delay restricted estimate of the channel,
wherein elements L+1 to N of the first channel estimation are suppressed, wherein L is a length of a channel impulse response and N is a number of sub-carriers in the received signal, and wherein N and L are integers greater than zero.

14. The receiver of claim 13, wherein the first matrix is unitary and is obtained from a singular value decomposition of a Fourier transform matrix.

15. The receiver of claim 13, further comprising an equalizer to equalize the received signal.

16. A non-transitory computer readable storage medium, comprising operational instructions that, when executed by a processor, cause the processor to:
perform a first channel estimation to obtain a first estimate of a channel based on a received signal;
suppress a portion of elements of the first estimate of the channel to obtain a suppressed estimate of the channel, wherein elements L+1 to N of the first channel estimation are suppressed, wherein L is a length of a channel impulse response and N is a number of sub-carriers in the received signal, and wherein N and L are integers greater than zero; and
multiply the suppressed estimate of the channel by a first matrix to obtain a delay restricted estimate of the channel.

17. The non-transitory computer readable storage medium of claim 16, wherein the first matrix is unitary.

18. The non-transitory computer readable storage medium of claim 17, wherein the unitary matrix is obtained from a singular value decomposition of a Fourier transform matrix.