Probability of Error, Digital Signaling on a Fading Channel And Equalization Schemes for ISI Wireless Communications Technologies Spring 2005 Lectures 11&12 R1 Department of Electrical Engineering, Rutgers University, Piscataway, NJ Instructor: Dr. Narayan Mandayam Summary by Randy Mitch Abstract – This document discusses the probability of error for non-coherent FSK and DPSK. Digital signaling on a frequency selective channel will be reviewed as well as equalization schemes to eliminate ISI. 1.0 Probability of Error for non-coherent BFSK Recall from the last lecture that we have Binary FSK orthogonal modulation. When a “1” is selected then S1(t) is transmitted and when a “0” is selected then S2(t)is transmitted. S1(t) and S2(t) are orthogonal. The received signal is g1(t) if S1(t) was transmitted and g2(t) if S2(t) is transmitted. The received signals g1(t) and g2(t) are also orthogonal. The receiver structure is shown in figure 1. Filter Matched to S1(t) t=T Envelope Detector l1 l1 l2 1 X(t) Compare l2 l1 0 Filter Matched to S2(t) Envelope Detector l2 t=T Figure 1. Non-coherent Receiver for BFSK Let S1(t) be transmitted, then an error occurs if l2 > l1. l2 xI 2 xQ2 Where XI2 and XQ2 are both Gaussian with zero mean and psd = N0/2. 2l2 l2 2 exp fl2 (l2 ) N 0 N0 0 , otherwise , l2 0 1 Pr obability of error Pe1 Pr l2 l1 Pr l2 1 l1 fl1 l1 dl1 0 l2 Pr l2 1 l1 fl2 l2 dl2 exp 1 N0 l1 N where l1 xI1 xQ1 , xI1 N E , 0 and xQ1 2 x E 2 I1 1 f xI xI1 exp N0 N0 2 1 xQ f xQ xQ1 exp 1 N0 N 0 N N 0, 0 2 Pr obability of error Pe1 p error xI1 xQ1 f xI xQ dxI1 dxQ1 1 1 exp x xQ1 2 xI1 2 xI1 E Pe1 exp exp N0 N0 Rewriting xQ1 2 xI1 2 xI1 E Pe1 2 xQ1 2 xI1 E 2 E 2 1 exp xI1 E exp N0 2 N N 0 0 2 2 2 xQ1 2 dxI dxQ N 0 1 1 2 Q1 E 2 2 xQ1 dx exp dxQ1 I1 N 0 E 1 Pe1 exp 2 2 N0 2.0 Probability of error for non-coherent M-ary FSK In general for M-ary FSK with non-coherent detection: Pe M i 1 kEb 1 i 1 exp 2 m 1 i 2 iN0 Where k log 2 M A non-coherent receiver for M-ary FSK is shown in Figure 2. As with the binary FSK receiver a comparison of the output of the envelope detectors selects the appropriate branch. Figure 3 illustrates the probability of bit error for various values of M. As M increases the bandwidth efficiency decreases but the power efficiency increases. More constellations require more chunks of orthogonal spaces. 2 Filter Matched to S1(t) t=T Envelope Detector t=T X(t) Filter Matched to S2(t) Envelope Detector Compare Filter Matched to Sm(t) t=T Envelope Detector Figure 2. Non-coherent Receiver for M-ary FSK Figure 3. Non-coherent M-ary FSK BER 3 3.0 Differential Phase Shift Keying If information is keyed onto the phase, PSK, it cannot be detected by non-coherent methods. Use phase difference between two consecutive waveforms to carry the information. Assumption: It works provided the unknown phase introduced by the channel varies slowly (i.e. slow enough to be considered constant over 2 bit intervals). Consider input sequence {mk}. Generate differentially encoded sequence {dk} from {mk} as follows: 1. Sum dk-1 and mk Modulo 2. 2. Set dk to be the complement of the result from 1 above. 3. Use dk to phase shift a carrier as follows: dk = 1 =0 dk = 0 = Example: if m k 10010011 dk 1 11011011 dk 10110111 0 00 000 Observe: Symbol d k is unchanged from previous symbol if incoming symbol is 1. Symbol d k is toggled from previous symbol if incoming symbol is 0. 4.0 - DQPSK 4 Exploit the above observation to derive the probability of error. Let the DPSK signal in Eb 0 t Tb be cos 2 f ct . If the next bit (in the interval Tb t 2 Tb) is 1, then the 2Tb phase is unchanged. If the next bit is 0, then the phase is shifted by . 4 S1 t Eb cos 2 f ct , 0 t Tb 2Tb Eb cos 2 f ct , Tb t 2Tb 2Tb S2 t Eb cos 2 f ct , 0 t Tb 2Tb Eb cos 2 f ct , Tb t 2Tb 2Tb It is possible to consider S1 t and S2 t as similar to non-coherent orthogonal modulation over a 2 bit interval, 0 t 2Tb. E 1 Pe exp 2 N0 Figure 4 compares the BER for DPSK and FSK. Observe that DPSK BER is approximately 3 dB better than non-coherent FSK. Figure 4 BER Comparisons of DPSK and FSK 5 5.0 Digital Signaling on Frequency Selective Fading Channels Modeling: Recall any modulated signal is given as v(t ) A b t kT , x k . We restrict k ourselves to linear modulation. b t , x k xk ha t xk complex symbol sequence amplitude sampling pulse The above signal is transmitted through a channel c(t) that results in the received signal w(t). w(t ) xk h t kT z (t ) k 0 zero-mean AWGN h(t ) h ( )c(t )d . a For causal channels h(t ) ha ( )c(t )d , t 0 . 0 We also assume h(t) = 0 for t 0 and h(t) = 0 for t LT. L is some real positive integer. x (t nT ) n k ha (t ) c (t ) u (t ) h* (t ) y (n ) t T z (t ) h (t ) Figure 5. Matched Filter Receiver in AWGN Channel If we know h(t), we can implement a matched filter as above. y (t ) x k k f (t kT ) v(t ) Filter noise 6 f (t ) h h( t )d . * f (t ) is the overall pulse response and it accounts for transient filter, channel and receive filters. yn y (nt ) x k k f (nT kT ) v(nT ) xk f n k vn k xn f 0 x k k n k f n 2 vn called intersymbol interference(ISI) ISI is caused by the channel so if we can eliminate it then we can treat the channel as AWGN channel. Therefore to achieve the same performance as on a AWGN channel, we require x k k n k f n 2 0 . In order to do this we must make f n2 = 0. Alternately, 0 if i j f k k0 f 0 where ij . 1 if i j Equally in the frequency domain 1 n F f f0 T n T , F ( f ) F f (t ) It can be shown that f(t) can be any function that has equally spaced zero crossings. What is the optimum receiver? We must express w(t ) lim wn n (t ) using the n Karhunen-Loeve expansion. T hnk h(t kT ) n* (t )dt 0 T zn z (t ) n* (t )dt 0 w w1 , w2 , ... wn is a multivariate Gaussian distribution because z(t) is Gaussian. 7 1 1 p ( w x, H ) exp wn xk hnk k n 1 N 0 N0 where H h1 , h 2 , h3 , ... ,hn and hn (...., hn3 , hn2 , hn1, h0 ) N 2 Therefore the optimum receiver is the Maximum Likelihood receiver, for the case of AWGN it reduces to N arg max u ( x) wn xk hnk x n 1 k 2 The bottom line is: 1. We need knowledge of f n to understand the channel response and knowledge of cn in order to perfectly eliminate ISI. Therefore we need to estimate the channel in order to equalize it. 2. An additional problem results as well by observing the following: x y (t ) k k f (t kT ) v(t ) where the noise function is h ( ) z (t )d v(t ) * is still Gaussian but is not white. Therefore the noise samples at the output are correlated. In this case we can obtain a discrete time white noise model as follows. x (t nT ) n k ha (t ) c (t ) h* (t ) Whitening Filter vk t T z (t ) h (t ) Figure 6. Whitening Filter with Matched Filter Receiver The output of the whitening filter vk is given by L vk g n xk n k n 0 8 where g n is the overall filter for the channel and the whitening filter and k is the white noise process. 3. Another important consideration is the design of ISI equalizing filters is extremely sensitive to timing information. We can solve this sensitivity in two ways. a. Use pulse shaping. Raised cosine pulse shaping allows us to derive the length of the pulse by sampling at even points. b. Use fractional sampling. Sample the output y(t) at a rate higher than 2/T. Although we will still have correlated noise and will need a whitening filter the overall pulse shape will be less sensitive to timing errors. 6.0 Equalization Schemes There are two types of equalization schemes. 1. Symbol by symbol equalizers that can be linear or non-linear. 2. Sequence estimation equalizers that are non-linear. 6.1 Symbol-by-Symbol Equalizers We consider the discrete time white noise model shown in figure 7. Where an is the input signal, L is the memory of the channel, n is the additive noise and g ( g0 , g1, .....,gl )T is the channel vector that describes the overall channel impulse response. an T g0 ……... T X g1 T X X gL L rn g k an k n k 0 n Figure 7. Discrete Time White Noise Model 9 Linear equalizers estimate the nth transmitted symbol aˆ n as aˆn M cr j n j j M Where c j M M is the equalizer filter of (2M+1) taps. M is a design choice and C is the chosen number of taps to estimate aˆ n . 6.2 Zero Forcing Equalizer The Zero Forcing Equalizer combines the channel and equalizer impulse response to force to zero at all but one of the taps in the TDL filter. The tap coefficients, c, are chosen to “zero-out” the ISI. Given the channel vector, g , we can select the tap coefficients, c, to get the desired response q (0, 0, 0,......, qn , 0,...) . The Zero Forcing Equalizer zeros out all but the desired result qn . The tap coefficients are calculated using the relationship qn c g n . Zero forcing equalizers can perfectly eliminate ISI as M but T it also enhances the noise. In practice, the receiver does not know the channel vector and finite length training sequences are used to choose the tap coefficients. 6.3 Minimum Mean Square Equalizer (MMSE) The MMSE is another symbol-by-symbol equalizer but is superior to the zero forcing equalizer in performance. The MSME utilizes the mean square error criterion to adjust the tap coefficients. We define an estimation error: n an aˆn Where an is the symbol sent and aˆ n is the estimated symbol. The function to be minimized is: J min E n 2 min E (an aˆn ) 2 c c 2 M min E (an c j rn j ) c j M The error is minimized by choosing c j so as to make the error sequence orthogonal to the signal sequence rn l , for l M , i.e. E n rnl 0 , l M . The optimum c is obtained by solving the following set of equations. M c E r j M j r E an rn l , n j n l l M 10 Define the following functions. and E rn j rnl , j l , M ,...., M b E an rn m E an rn m 1 ........E an rn m then copt 1 b To implement the MSME we need to us adaptive algorithms because the channel response is not readily known. Typically, we use the steepest descent algorithm. 2 1 E n c j (n 1) c j (n) , j M ....... M 2 c j Where is a positive number. Note: E 2 n (n) 2 E (n) 2 E (n)rn j 2 Rer ( j ) c j c j The term Rer is the cross correlation between the input and the error. The equalizer taps can be obtained by implementing a stochastic gradient algorithm. 1 c j (n 1) c j (n) (n)rn j 2 To evaluate (n) in this algorithm we use a training sequence. 6.4 Decision Feedback Equalizer (DFE) A DFE is a nonlinear equalizer that consists of a feed forward section and a feedback section. DFEs are very effective in frequency selective channels because they mitigate the effects of noise enhancements that degrade the performance of linear equalizers. The DFE uses previous decisions to eliminate ISI caused by previously detected symbols on current symbols. The output of the equalizer can be expressed as: aˆn 0 j M1 c j rn j feed forwad filter M1 1 taps M2 c j an j j 1 feedback filter M 2 taps Where an 1 , an 2 , ....., an M 2 are earlier detected symbols. Both the feed forward loop and the feedback loop need adaptation to adjust the coefficients. 11 6.5 Maximum Likelihood Sequence Estimation (MLSE) Recall the discrete-time white noise channel model. L rn g m an m n m 0 Assume k symbols are transmitted over the channel. Then after receiving the sequence k k rn n 1 , the ML receiver decides in favor of the sequence an n 1 that maximizes the likelihood function log rk , rk 1, ..., r1 ak , ak 1 ,..., a1 Since the noise samples are independent and rn depends only on the L most recent transmitted symbols. log rk , rk 1, ..., r1 ak , ak 1 ,..., a1 log rk ak , ak 1 ,., ak l log rk 1 , rk 2, .., r1 ak 1, ak 2 ,.., a1 This has already been calculated where ak L 0 for k L 0 Since the 2nd term has been calculated at the previous time (k-1), only the first term needs to be computed at time k for each incoming signal rk . The Veterbi algorithm can be used to implement the MLSE equalizer. Adaptive MLSE is used in GSM. References: [1] [2] [3] [4] [5] [6] N. Mandayam, Wireless Communication Technologies, course notes. T. Rappaport, Wireless Communications. Principles and Practice. 2nd Edition, Prentice-Hall, Englewood Cliffs, NJ: 1996. J. Proakis, Digital Communications, 3rd Edition, McGraw-Hill, NY: 1995. G. Stuber, Principles of Mobile Communication, 2nd Edition, Kluwer Academic Publishers, Norwell, MA: 2001. B. Sklar, Digital Communications. Fundamentals and Applications, Prentice-Hall, Englewood Cliffs, NJ: 1988. D. Wu, Wireless Communication Technologies, Rutgers University, Piscataway, NJ: 2002 12