Abstract:
A digital signal processor which performs digital signal processing of a digital signal includes a statistical analysis method which calculates a moving average and a standard deviation from the digital signal, performs statistical decision deciding whether or not the digital signal is within a predetermined range obtained from the moving average and the standard deviation, and corrects the digital signal outside the range within the range. Statistical analysis of the digital signal is performed, thereby suppressing transient changes without increasing the number of times of averaging during the digital signal processing.

Description:
CROSS REFERENCE TO RELATED APPLICATIONS 
     This application is a U.S. National Stage application claiming the benefit of prior filed International Application Number PCT/JP2013/004398, filed on Jul. 18, 2013, in which the International Application claims priority from Japanese Patent Application Number 2012-195209, filed on Sep. 5, 2012, the entire contents of which are incorporated herein by reference. 
     TECHNICAL FIELD 
     The present application relates to a digital signal processor which suppresses transient changes in a digital signal. 
     BACKGROUND ART 
     In a digital signal processor, an analog signal is converted to a digital signal by an AD (Analog to Digital) converter, and digital signal processing is performed on the digital signal, whereby, for example, a received waveform distorted in a transmission line can be compensated in a digital region. The digital signal processor is used in various technical fields, such as image processing, sound processing, wireless communication, and optical communication. 
     When performing the digital signal processing, if transient changes, such as pulse noise, occur in the digital signal during the digital signal processing, errors in the digital signal processing increase and the quality of an output signal is deteriorated. However, the influence of the transient change components on the digital signal processing can be reduced by, for example, averaging processing of the digital signal using a low pass filter. 
     For example, in a digital coherent receiver in an optical communication field, in order to reduce the influence of the transient changes, there is a method which, when performing equalization processing on a received waveform using a FIR (Finite Impulse Response) filter, increases the number of taps of FIR filters. Alternatively, there is a method which estimates the phase shift from the phase point of the phase modulation signal using an M-power algorithm, and when removing the phase shift, increases the cumulative number of signals (the number of taps) of the M-power algorithm. 
     Here, a digital coherent receiver which uses a coherent optical communication technique and a digital signal processing technique in combination will be described. 
       FIG. 5  shows a configuration example of a digital coherent transmission/reception system (Non-Patent Document 1). 
     In  FIG. 5 , the digital coherent transmission/reception system has a transmitter  100  which transmits a phase-modulated optical signal, and a digital coherent receiver  200  which receives and demodulates the optical signal transmitted through a transmission line. The digital coherent receiver  200  has a coherent receiver  210 , an AD converter  220 , and a digital signal processor  230 . The coherent receiver  210  inputs the optical signal received from the transmission line and local light from an optical local oscillator  301  and converts the optical signal to an electrical signal by a coherent detection technique with high sensitivity. The AD converter  220  converts the electrical signal output from the coherent receiver  210  to a digital signal. The digital signal processor  230  performs digital signal processing on the digital signal output from the AD converter  220  and demodulates the digital signal while compensating for a received waveform distorted in the transmission line. 
     The digital signal processor  230  has an equalizer  231 , a phase shift compensator  232 , and a demodulator  233 . The equalizer  231  equalizes waveform distortion of the input digital signal, and the phase shift of the waveform-equalized digital signal is compensated by the phase shift compensator  232 . The demodulator  233  outputs the phase shift-compensated digital signal output from the phase shift compensator  232  as a symbol string. In this way, since the correction of the waveform distortion can be performed with a simple configuration, a large-capacity and high-speed transmission system can be realized. 
     The phase shift compensator  232  can estimate and correct the phase shift using, for example, the M-power algorithm (Non-Patent Document 2). Since the estimation range of the phase shift in the M-power algorithm is limited within the range of ±π/4 from a reference point for a QPSK (Quadrature Phase Shift Keying) signal, a phase shift outside the range cannot be estimated. A phenomenon in which the time continuity of the phase shift estimation values is not maintained is called a “cycle slip”, and signal quality is deteriorated. For example, when transient changes like pulse noise occur in the digital signal, time continuity is not maintained due to the error expansion of the digital signal processing, and as shown in  FIG. 6 , a cycle slip occurs. 
     As a countermeasure against the cycle slip, a method which performs logical differential coding on a transmission signal to prevent the propagation of the influence, or the like is used (Non-Patent Document 3). However, bit errors at the moment when the cycle slip occurs cannot be prevented. When one bit error occurs in differentially coded data, since the bit error is subjected to differential decoding as continuous two bit errors, transmission quality is deteriorated.
     Non-Patent Document 1: S. J. Savory, “Digital filters for coherent optical receivers” Optics Express, vol. 16, no. 2, pp. 804-814, 2008   Non-Patent Document 2: S. Tsukamoto, Y. Ishikawa, and K. Kikuchi, “Optical Homodyne Receiver Comprising Phase and Polarization Diversities with Digital Signal Processing” Proc. ECOC, 2006   Non-Patent Document 3: T. Mizuochi, Y. Miyata, K. Kubo, T. Sugihara, K. Onohara and H. Yoshida, “Progress in Soft-Decision FEC” OSA/OFC/NFOEC, NWC2, 2011   

     DISCLOSURE 
     Problems to be Solved 
     While it is possible to cope with the transient changes of the digital signal, such as the phase shift from the phase point of the phase modulation signal, by performing averaging of the digital signal, averaging causes an increase in delay amount during signal processing, degradation of followability, or the like. 
     A proposition of the present application is to provide a digital signal processor capable of suppressing transient changes of a digital signal by performing statistical analysis of the digital signal without increasing the number of times of averaging during digital signal processing. 
     Means for Solving the Problems 
     The present application provides a digital signal processor which performs digital signal processing of a digital signal, the digital signal processor including a statistical analysis method calculating a moving average and a standard deviation from the digital signal, performing statistical decision deciding whether or not the digital signal is within a predetermined range obtained from the moving average and the standard deviation, and correcting the digital signal outside the range to be within the range. 
     In the digital signal processor of the present application, the statistical analysis method includes a moving average calculation block which inputs the digital signal and outputs the moving average, a standard deviation calculation block which inputs the digital signal and the moving average output from the moving average calculation block and outputs the standard deviation, and a statistical decision/signal correction block which inputs the digital signal, the moving average, and the standard deviation, corrects the digital signal by the statistical decision for the digital signal, and outputs the digital signal being corrected. 
     In the digital signal processor of the present application, when n is an integer equal to or greater than 3 and L is an integer equal to or greater than 2, the moving average calculation block inputs digital signals of L points in total from an (n−L)th digital signal S(n−L) to an (n−1)th digital signal S(n−1) and outputs a moving average A(n−1); the standard deviation calculation block inputs the digital signals of the L points to the (n−1)th digital signal and the moving average A(n−1) output from the moving average calculation block and outputs a standard deviation σ(n−1); and the statistical decision/signal correction block inputs an n-th digital signal S(n), the moving average A(n−1) output from the moving average calculation block, and the standard deviation σ(n−1) output from the standard deviation calculation block, performs the statistical decision deciding, with an arbitrary positive number as x, whether or not the digital signal S(n) is within a range of:
 
 A ( n− 1)− x σ( n− 1)≦ S ( n )≦ A ( n− 1)+ x σ( n− 1)
 
outputs the digital signal S(n) as it is when the digital signal S(n) is within the range, and corrects the digital signal S(n) to be within the range when the digital signal S(n) is outside the range and outputs the corrected digital signal S(n).
 
     Effect 
     The present application corrects a digital signal by statistical decision for the digital signal during digital signal processing, thereby suppressing transient changes of the digital signal and improving stability of the digital signal processing. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  is a diagram showing a configuration example of a digital signal processor of the present application. 
         FIG. 2  is a diagram showing a configuration example of a statistical analysis method  234 . 
         FIG. 3  is a diagram showing a processing example of the statistical analysis method  234  which corrects a phase shift. 
         FIG. 4  is a diagram showing a time-varying phase shift example in the present application. 
         FIG. 5  is a diagram showing a configuration example of a digital coherent transmission/reception system. 
         FIG. 6  is a diagram showing a time-varying phase shift example in a conventional configuration. 
     
    
    
     DETAILED DESCRIPTION OF THE EMBODIMENTS 
       FIG. 1  shows a configuration example of a digital signal processor of the present invention. Here, although an example where a digital signal processor is applied to a digital coherent receiver  200  of a digital coherent transmission/reception system shown in  FIG. 5  will be described, the invention is not limited to a digital signal processor  230  of the digital coherent receiver  200 . In the present embodiment, although a technique for suppressing transient changes of a digital signal, such as the phase shift from the phase point of the phase modulation signal, will be described, for example, the invention can be similarly applied to general digital signal processing for suppressing transient noise during sound signal processing. 
     In  FIG. 1 , a transmitter  100 , a digital coherent receiver  200 , an optical local oscillator  301 , a coherent receiver  210 , an AD converter  220 , and a digital signal processor  230  of the digital coherent receiver  200 , and an equalizer  231 , a phase shift compensator  232 , and a demodulator  233  of the digital signal processor  230  have the same functions as the configuration shown in  FIG. 5 . 
     The digital signal processor  230  of the present embodiment has a feature in which a statistical analysis method  234  is connected to the phase shift compensator  232 , thereby suppressing transient changes of a digital signal, such as the phase shift from the phase point of the phase modulation signal, and securing time continuity of the phase modulation signal. That is, a digital signal which exhibit a phase shift output from the phase shift compensator  232  are input to the statistical analysis method  234 , and the statistical analysis method  234  returns the digital signal with the uncorrected or corrected phase shift by statistical analysis processing described below to the phase shift compensator  232 . The phase shift compensator  232  compensates for the phase shift of the digital signal input from the equalizer  231  using the digital signal with the uncorrected or corrected phase shift input from the statistical analysis method  234  and outputs the compensated digital signal to the demodulator  233 . 
       FIG. 2  shows a configuration example of the statistical analysis method  234 . 
     In  FIG. 2 , the statistical analysis method  234  includes a moving average calculation block  11  which calculates a moving average of the input a digital signal, a standard deviation calculation block  12  which calculates a standard deviation from the input digital signal and the moving average, and a statistical decision/signal correction block  13  which performs statistical decision for the digital signal described below using the moving average and the standard deviation of the digital signal and outputs the digital signal with the uncorrected or corrected phase shift. 
     The moving average calculation block  11  inputs digital signals of L points in total from an (n−L)th digital signal S(n−L) to an (n−1)th digital signal S(n−1) when n is an integer equal to or greater than 3 and L is an integer equal to or greater than 2, and calculates a moving average A(n−1). 
     The standard deviation calculation block  12  inputs the digital signals of the L points from the (n−L)th digital signal to the (n−1)th digital signal and the moving average A(n−1) output from the moving average calculation block  11  and calculates a standard deviation σ(n−1). 
     The statistical decision/signal correction block  13  inputs the n−th digital signal S(n), the moving average A(n−1) output from the moving average calculation block  11 , and the standard deviation σ(n−1) output from the standard deviation calculation block  12 , performs the statistical decision deciding, with an arbitrary positive number as x, whether or not the digital signal S(n) is within a range of:
 
 A ( n− 1)− x σ( n− 1)≦ S ( n )≦ A ( n− 1)+ x σ( n− 1)
 
     outputs the phase shift of the digital signal S(n) as it is when the digital signal S(n) is within the range, and corrects the phase shift of the digital signal S(n) to be within the range and outputs the corrected phase shift of the digital signal S(n) when the digital signal S(n) is outside the range. 
     Here, when x=2, the above-described range is:
 
 A ( n− 1)−2σ( n− 1)≦ S ( n )≦ A ( n− 1)+2σ( n− 1)
 
     For example, when the digital signal S(n) is smaller than A(n−1)−2σ(n−1), it is corrected to:
 
 S ( n )= A ( n− 1)−2σ( n− 1)
 
     and when the digital signal S(n) is greater than A(n−1)+2σ(n−1), it is corrected to:
 
 S ( n )= A ( n− 1)+2σ( n− 1)
 
     With this, it is possible to remove the transient changes of the digital signal. 
     A moving average A(m) and a standard deviation σ(m) of the digital signals of the L points to the m-th digital signal can be respectively calculated by the following expressions. 
     
       
         
           
             
               
                 
                   
                     
                       A 
                       ⁡ 
                       
                         ( 
                         m 
                         ) 
                       
                     
                     = 
                     
                       
                         1 
                         L 
                       
                       ⁢ 
                       
                         
                           ∑ 
                           
                             k 
                             = 
                             
                               m 
                               - 
                               L 
                               + 
                               1 
                             
                           
                           m 
                         
                         ⁢ 
                         
                             
                         
                         ⁢ 
                         
                           S 
                           ⁡ 
                           
                             ( 
                             k 
                             ) 
                           
                         
                       
                     
                   
                   ⁢ 
                   
                     
 
                   
                   ⁢ 
                   
                     
                       σ 
                       ⁡ 
                       
                         ( 
                         m 
                         ) 
                       
                     
                     = 
                     
                       
                         
                           1 
                           L 
                         
                         ⁢ 
                         
                           
                             ∑ 
                             
                               k 
                               = 
                               
                                 m 
                                 - 
                                 L 
                                 + 
                                 1 
                               
                             
                             m 
                           
                           ⁢ 
                           
                               
                           
                           ⁢ 
                           
                             
                               ( 
                               
                                 
                                   S 
                                   ⁡ 
                                   
                                     ( 
                                     k 
                                     ) 
                                   
                                 
                                 - 
                                 
                                   A 
                                   ⁡ 
                                   
                                     ( 
                                     m 
                                     ) 
                                   
                                 
                               
                               ) 
                             
                             2 
                           
                         
                       
                     
                   
                 
               
               
                 
                   [ 
                   
                     Equation 
                     ⁢ 
                     
                         
                     
                     ⁢ 
                     1 
                   
                   ] 
                 
               
             
           
         
       
     
       FIG. 3  shows a processing example of the statistical analysis method  234  which corrects the phase shift. Here, a processing example of the statistical analysis method  234  in a range of 0.750 μs to 0.765 μs in  FIG. 6  with L=15 and x=2 is shown. 
     A point B of  FIG. 3  is outside the range of “15-period moving average−2σ” of a point A due to a transient change of the phase shift, continuity of the phase shift is not maintained by error expansion of subsequent signal processing, and a cycle slip occurs. The point B is corrected to be within the range of “15-period moving average±2σ” at the point A in the above-described manner to suppress the transient change, whereby, as shown in  FIG. 4 , time continuity of the phase shift can be secured. In  FIG. 3 , a phase shift outside the range of “15-period moving average±2σ” like a point C subsequent to the point B is within the range of “15-period moving average±2σ” by correcting the point B to be within the range of “15-period moving average±2σ” at the point A. 
     In  FIG. 4 , a signal has a bit error rate (BER)=1.9×10 −3 , and signal quality is improved compared to BER=1.4×10 −2  when a cycle slip occurs. 
     The present invention can be applied to various kinds of digital signal processing for a time-varying digital signal as well as the time-varying phase shift in the phase shift compensator  232  of the digital signal processor  230  of the digital coherent receiver  200  shown in  FIG. 1 . For example, the present invention can be applied to suppressing transient changes during adaptive control of the tap coefficient of a FIR filter in the equalizer  231 . The present invention can be applied to suppressing transient noise during sound signal processing. 
     The many features and advantages of the embodiments are apparent from the detailed specification and, thus, it is intended by the appended claims to cover all such features and advantages of the embodiments that fall within the true spirit and scope thereof. Further, since numerous modifications and changes will readily occur to those skilled in the art, it is not desired to limit the inventive embodiments to exact construction and operation illustrated and described, and accordingly all suitable modifications and equivalents may be resorted to, falling within the scope thereof.