Pilot symbol-assisted coherent multistage interference cancellerusing recursive channel estimation for DS-CDMA mobile radio

A pilot symbol-assisted coherent multistage interference canceller using recursive channel estimation is proposed for DS-CDMA mobile radio cellular system. Since the channel variation caused by fading is recursively estimated at each interference cancelling stage, the accuracy of channel estimation is improved successively. Computer simulation results show that the required E_b/N₀ at the average BER of 3×10^-2 is improved by ~3.5 dB compared to the matched filter receiver for 10 users and two paths with equal power, and where f_dT=10^-3 (f_d: fading maximum Doppler frequency, T: data symbol duration)     

Experimental evaluation of combined effect of coherent RAKEcombining and SIR-based fast transmit power control for reverse link ofDS-CDMA mobile radio

The combined effect of coherent RAKE combining using the weighted multislot averaging (WMSA) channel estimation filter and closed-loop fast transmit power control (TPC) in the 4.096 Mchip/s direct sequence code division multiple access (DS-CDMA) mobile radio reverse link is experimentally evaluated. The WMSA channel estimation filter utilizes periodically transmitted pilot symbols (four pilot symbols are time-multiplexed in each 40-symbol time slot). Its observation period is extended to 2-K slots in order to improve the accuracy of the channel estimation. The fast TPC is based on the measurement of signal-to-interference plus background noise ratio (SIR) using pilot symbols. Laboratory experiments show that the use of the K=2 WMSA channel estimation filter reduces the required E_b/I₀ at the average BER of 10^-3 by approximately 0.5 dB compared to use of the linear interpolation filter, and that the required E_b/I₀ is minimized when the SIR measurement interval is M=10 symbols (one slot TPC delay). It was also clarified that SIR-based TPC works satisfactorily when two users with different information data rates, i.e., SF, independently employ fast TPC. Field experimental results obtained in an area nearby Tokyo showed that the average BER of 10^-3 is achieved at the target E_b/I₀ per antenna of approximately 2.5 dB by using four-finger branch RAKE and two-branch antenna diversity. Although the target E_b/I₀ to achieve same BER, when there is one interfering user with a fourfold greater transmit power than that of the desired user that independently employs fast TPC, is almost the same as that in the single-user case, the mobile transmit power is increased by 1.0-2.0 dB due to the increased MAI. These results indicate that the combination of coherent RAKE combining and fast TPC works well in practical multipath fading channels     

Mixed H2/H∞ filtering design inmultirate transmultiplexer systems: LMI approach

A mixed H₂/H_∞ filter design is proposed for multirate transmultiplexer systems with dispersive channel and additive noise. First, a multirate state-space representation is introduced for the transmultiplexer with the consideration of channel dispersion. Then, the problem of signal reconstruction can be regarded as a state estimation problem. In order to design an efficient separating filterbank for a transmultiplexer system with uncertain input signal and additive noise, the H_∞ filter is employed for robust signal reconstruction. The H₂ filter design is considered to be a suboptimal approach to achieve the optimal signal reconstruction in transmultiplexer system under unitary noise power. Finally, a mixed H₂/H_∞ filter is proposed to achieve a better signal reconstruction performance in transmultiplexer systems. These design problems can be transformed to solving the eigenvalue problems (EVP) under some linear matrix inequality (LMI) constraint. The LMI Matlab toolbox can be applied to efficiently solve the EVP by convex optimization technique     

10 GHz dual-conversion low-IF downconverter with microwave and analogue quadrature generators

A 10 GHz dual-conversion low-IF downconverter using 0.18-mum CMOS technology is demonstrated. The high-frequency quadrature RF and LO₁ signals are generated by broadside-coupled quadrature couplers while a two-section polyphase filter is utilised for the low-frequency LO₂ quadrature signal generation. As a result, the demonstrated downconverter achieves a conversion gain of 7 dB, IP_{1 dB} of -16 dBm, IIP₃ of -5 dBm and noise figure of 26 dB at a 1.8 V supply. The image-rejection ratio of the first/second image signal is 33/42 dB for IF frequency ranging from 10 to 60 MHz, respectively.     

Deconvolution filter design via l1 optimizationtechnique

A new l₁ optimal deconvolution filter design approach for systems with uncertain (or unknown)-but-bounded inputs and external noises is proposed. The purpose of this deconvolution filter is to minimize the peak gain from the input signal and noise to the error by the viewpoint of the time domain. The solution consists of two steps. In the first step, the l₁ norm minimization problem is transferred to an equivalent A-norm minimization problem, and the minimum value of the peak gain is calculated. In the second step, based on the minimum peak gain, the l₁ optimal deconvolution filter is constructed by solving a set of constrained linear equations. Some techniques of inner-outer factorization, polynominal spectral factorization, linear programming, and some optimization theorems found in a book by Luenberger are applied to treat the l₁ optimal deconvolution filter design problem. Although the analysis of the algorithm seems complicated, the calculation of the proposed design algorithm for actual systems is simple. Finally, one numerical example is given to illustrate the proposed design approach. Several simulation results have confirmed that the proposed l₁ optimal deconvolution filter has more robustness than the l₂ optimal deconvolution filter under uncertain driving signals and noises     

Novel approach to the design of I/Q demodulation filters

A novel filter design approach to digital I/Q demodulation is proposed. Two possible realisations are presented using this approach. The first one is based on the highpass filter method which is suitable for B⩽f₀ and while the other realisation is based on the lowpass filter method suitable for B⩽f₀, where B and f ₀ are the IF signal bandwidth and the IF frequency, respectively. Both new realisations maintain the advantages of an earlier lowpass approach such as zero DC offset, matched channel frequency responses, and good performance over a wide bandwidth. At the same time, the new highpass filter realisation method possesses higher computational efficiency than other wideband approaches reported in the literature     

Short wavelets and matrix dilation equations

Scaling functions and orthogonal wavelets are created from the coefficients of a lowpass and highpass filter (in a two-band orthogonal filter bank). For “multifilters” those coefficients are matrices. This gives a new block structure for the filter bank, and leads to multiple scaling functions and wavelets. Geronimo, Hardin, and Massopust (see J. Approx. Theory, vol.78, p.373-401, 1994) constructed two scaling functions that have extra properties not previously achieved. The functions Φ₁ and Φ₂ are symmetric (linear phase) and they have short support (two intervals or less), while their translates form an orthogonal family. For any single function Φ, apart from Haar's piecewise constants, those extra properties are known to be impossible. The novelty is to introduce 2×2 matrix coefficients while retaining orthogonality of the multiwavelets. This note derives the properties of Φ₁ and Φ₂ from the matrix dilation equation that they satisfy. Then our main step is to construct associated wavelets: two wavelets for two scaling functions. The properties were derived by Geronimo, Hardin, and Massopust from the iterated interpolation that led to Φ1 and Φ₂. One pair of wavelets was found earlier by direct solution of the orthogonality conditions (using Mathematica). Our construction is in parallel with recent progress by Hardin and Geronimo, to develop the underlying algebra from the matrix coefficients in the dilation equation-in another language, to build the 4×4 paraunitary polyphase matrix in the filter bank. The short support opens new possibilities for applications of filters and wavelets near boundaries     

Iterative channel estimation and decoding of pilot symbol assistedturbo codes over flat-fading channels

A method for coherently detecting and decoding turbo-coded binary phase shift keying (BPSK) signals transmitted over frequency-flat fading channels is discussed. Estimates of the complex channel gain and variance of the additive noise are derived first from known pilot symbols and an estimation filter. After each iteration of turbo decoding, the channel estimates are refined using information fed back from the decoder. Both hard-decision and soft-decision feedback are considered and compared with three baseline turbo-coded systems: (1) a BPSK system that has perfect channel estimates; (2) a system that uses differential phase shift keying and hence needs no estimates; and (3) a system that performs channel estimation using pilot symbols but has no feedback path from decoder to estimator. Performance can be further improved by borrowing channel estimates from the previously decoded frame. Simulation results show the influence of pilot symbol spacing, estimation filter size and type, and fade rate. Performance within 0.49 and 1.16 dB of turbo-coded BPSK with perfect coherent detection is observed at a bit-error rate of 10^-4 for normalized fade rates of f_dT_s=0.005 and f_dT_s=0.02, respectively     

Multiple Adaptive Frequency Filtering for OFDM Channel Estimation

In orthogonal frequency division multiplexing (OFDM) systems using pilots, channel estimation is performed on pilots and then interpolated over the time and the frequency axis. As time-interpolated estimates have a different mean square error than pilot estimates we propose the use of multiple adaptive filters for exploiting the frequency correlation of the channel. Each filter operates on a different configuration of pilot and time-interpolated estimates and is adapted independently from the other filters. As a simpler suboptimal solution we consider also the use of a single filter, whose adaptation however takes into account the reliability of the various estimates. Performance results are reported with reference to the DVB-T and DVB-T2 standards showing that the proposed technique performs similar or better than existing approaches at a much lower complexity.      

Robust ℋ∞ filtering for continuous timevarying uncertain systems with deterministic input signals

Many dynamical systems involve not only process and measurement noise signals but also parameter uncertainty and known input signals. When ℒ₂ or ℋ_∞ filters that were designed based on a “nominal” model of the system are applied, the presence of parameter uncertainty will not only affect the noise attenuation property of the filter but also introduce a bias proportional to the known input signal, and the latter may be very appreciable. We introduce a finite-horizon robust ℋ_∞ filtering method that provides a guaranteed ℋ_∞ bound for the estimation error in the presence of both parameter uncertainty and a known input signal. This method is developed by using a game-theoretic approach, and the results generalize those obtained for cases without parameter uncertainty or without a known input signal. It is also demonstrated, via an example, that the proposed method provides significantly improved signal estimates     

Influence of Frequency Allocations and Optical Filters on FDM Optical Fiber Communications

In long-haul frequency-division-multiplexing (FDM) lightwave transmission systems, transmission characteristics are degraded by four-wave mixing (FWM) generated in optical fibers. To date, equally spaced (ES), unequally spaced (US), and repeated US (RUS) allocations have been demonstrated in FDM lightwave transmission systems. It has already been theoretically and experimentally revealed that intensities of generated FWM lights with f_FWM=f_i for RUS are lower than those for ES, and a total bandwidth of signal lights for RUS is narrower than that for US, where f_FWM is a frequency of an FWM light, and f_i is a frequency of a signal light with a channel index i. Moreover, to overcome RUS, ES RUS (ERUS) and US RUS (URUS) have been proposed as modified RUSs and theoretically analyzed. In this paper, ES, RUS, ERUS, and URUS allocations are studied from the viewpoint of a transmission bandwidth of an optical filter. It is revealed that FWM noises are reduced in URUS more than in ES, RUS, and ERUS with a decrease in a transmission bandwidth of an optical filter. The upper limit of a filter transmission bandwidth to achieve a bit error rate of 10^-9 is also obtained.     

Design techniques for tunable transresistance-C VHF bandpassfilters

In this paper, a tunable wideband linear transresistance (R_m ) amplifier is proposed and analyzed. Using the tunable R_m amplifier, a new transresistance-capacitor (R_m-C) differentiator is designed. Considering the intrinsic capacitances of the MOS transistors as filter elements, this R_m-C configuration can he regarded as a very high frequency (VHF) bandpass biquadratic filter. The proposed biquad has a simple structure and thus occupies a small chip area and consumes little power. Moreover, higher-order VHF bandpass filters can be realized by directly cascading the biquads. Experimental results have successfully proven the capability of the proposed new filter implementation method in realizing VHF bandpass filters with the center frequency higher than 100 MHz when C_d=1 pF. The deviations of the measured center frequency f _o and quality factor Q of the fabricated bandpass filter from the simulated results are less than 8%. The deviation of the center frequency can be post-tuned by adjusting the control voltages V_CN and V_CP of the tunable R_m amplifier. With 1 pF differentiating capacitor, the center frequency of the fabricated VHF R_m-C bandpass filters can be tuned in a wide range larger than 30 MHz. The measured maximum signal level is 25 mV_rms and the dynamic range is 47 dB. The chip area is 0.05 mm² and power consumption is 5.05 mW with ±2.5 V power supply     

Multipath interference canceller for high-speed packet transmissionwith adaptive modulation and coding scheme in W-CDMA forward link

This paper proposes a multipath interference canceller (MPIC) associated with orthogonal code-multiplexing that achieves much higher peak throughput than 2 Mbit/s with adaptive data modulation for high-speed packet transmission in the wideband direct sequence-code division multiple access (W-CDMA) forward link, and evaluates its throughput performance by computer simulation. The simulation results elucidate that sufficient multipath interference (MPI) suppression is achieved by a four-stage MPIC with 6-12 orthogonal code-multiplexing using one iterative channel estimation with pilot and decision feedback data symbols and further that the interference rejection weight control according to the number of observed multipaths is effective in improving the throughput. It is also demonstrated that MPIC exhibits a superior MPI suppression effect to a chip equalizer in the lower received signal energy per bit-to-background noise spectrum density (E_b/N₀) channel around 0-3 dB owing to the successive channel estimation at each stage. We show that the maximum peak throughput using MPIC is approximately 2.1 fold that without MPIC in a two-path and three-path Rayleigh fading channel and that the peak throughput of 8.0 Mbit/s is achieved using 64 QAM data modulation in a two-path fading channel within a 5 MHz bandwidth. Furthermore, the required average E_b/N₀ for satisfying the same throughput with MPIC is decreased by more than 2.0 dB     

Design of efficient M-band coders with linear-phase andperfect-reconstruction properties

This paper presents a new design technique for obtaining M-band orthogonal coders where M=2ⁱ. The structures obtained using the proposed technique have the perfect reconstruction property. Furthermore, all filters that constitute the subband coder are linear-phase FIR-type filters. In contrast with conventional design techniques that attempt to find a unitary alias-component matrix in the frequency domain, we carry out the design in the time domain, based on time-domain orthonormality constraints that the filters must satisfy. The M-band design problem is reduced to the problem of finding a suitable lowpass filter h₀(n). Once a suitable lowpass filter is found, the remaining (M-1) filters of the coder are obtained through the use of shuffling operators on the lowpass filter. This approach leads to a set of filters that use the same numerical coefficient values in different shift positions, allowing very efficient numerical implementation of the subband coder. In addition, by imposing further constraints on the lowpass branch impulse response h₀(n), we are able to construct continuous bases of M-channel wavelets with good regularity properties. Design examples are presented for four-, eight-, and 16-band coders, along with examples of continuous wavelet bases that they generate     

Simple coherent receivers for partial response continuous phasemodulation

By using a pulse-amplitude-modulation representation of binary continuous-phase-modulation signals, the authors develops a novel optimum Viterbi sequence detector and a near-optimum Viterbi receiver with low complexity. For modulation index 0.5, where a linear receiver can be used, a minimum-mean-squared-error linear receiver filter is derived. The performance of all of these is analyzed, using the Gaussian minimum-shift-keying signal (GMSK) for illustration. It is shown that a GMSK receiver consisting of two matched filters and a four-state Viterbi algorithm performs with less than 0.24-dB degradation compared with the optimal receiver. The linear receiver is optimum for all values of E _b/N₀ (bit-energy-to-noise one-sided spectral density ratio). A design method for its filter is given. The filter is equivalent to a cascade of a matched filter and a Wiener filter estimator. Both upper and lower bounds for the bit-error probability are calculated. Simulation results which confirm the analysis are given     

A direct approach to H2 optimal deconvolution ofperiodic digital channels

This paper studies the H₂ optimal deconvolution problem for periodic finite impulse response (FIR) and infinite impulse response (IIR) channels. It shows that the H₂ norm of a periodic filter can be directly quantified in terms of periodic system matrices and linear matrix inequalities (LMIs) without resorting to the commonly used lifting technique. The optimal signal reconstruction problem is then formulated as an optimization problem subject to a set of matrix inequality constraints. Under this framework, the optimization of both the FIR and IIR periodic deconvolution filters can be made convex, solved using the interior point method, and computed by using the Matlab LMI Toolbox. The robust deconvolution problem for periodic FIR and IIR channels with polytopic uncertainties are further formulated and solved, also by convex optimization and the LMIs. Compared with the lifting approach to the design of periodic filters, the proposed approach is simpler yet more powerful in dealing with multiobjective deconvolution problems and channel uncertainties, especially for IIR deconvolution filter design. The obtained solutions are applied to the design of an optimal filterbank yielding satisfactory performance     

Path diversity for FFH/PSK spread-spectrum communication systems

A fast frequency hopping (FFH) method which uses path-diversity combining is proposed. Diversity techniques are realized when a signal is received from diverse independent paths, each of which carries identical information but suffers from independent fading variations. The improvement of communication performance of FFH systems is possible as the delayed paths are used and path-diversity combination is realized. The advantages of this method, operating in Rayleigh fading channels are confirmed by theoretical analyses. The improvement is in order of 2~3 dB at bit error rate (BER) of 10^-3. This method can be also effective against interferences from other users in a code division multiple access (CDMA) environment. The performance of this system in a CDMA environment is evaluated by theoretical analyses and is shown to be superior to non-combining method. At BER of 10^-3 the required E_b/N₀ of the proposed system is 5 dB lower. If E_b/N_o is fixed, a greater number of users can be accommodated using the proposed system     

A proposal of broad-bandwidth vertical-cavity laser amplifier

We propose a generic vertical-cavity amplifier (VCA) using a coupled-cavity design to broaden the bandwidth. Calculations are made for cavities with GaAs-AlAs and GaAs-Al_xO₂ distributed Bragg reflectors (DBR). We found that at reasonable pumping levels the VCA increased the bandwidth by 85% (GaAs-AlAs) to 500% (GaAs-Al_xO₂) as compared to a simple two-mirror structure similar to vertical-cavity surface-emitting lasers. In particular, the GaAs-AlAs VCA shows a bandwidth of 2 nm at 6-dB signal gain, while the GaAs-Al_xO₂ VCA demonstrates a 5-nm bandwidth at 6-dB signal gain with no ripple at required single-pass power gain of ~2-2.5%. Furthermore, as large as 30-nm bandwidth in a lossless bandpass filter can be obtained     

Theoretical analysis of word-level switching activity in thepresence of glitching and correlation

This paper presents a novel analytical approach to compute the switching activity in digital circuits at the word level in the presence of glitching and correlation. The proposed approach makes use of signal statistics such as mean, variance, and autocorrelation. It is shown that the switching activity α_f at the output node f of any arbitrary circuit in the presence of glitching and correlation is computed as α_f=Σ_i=1^S-1α(f _i,i+1)=Σ_i=1^{S- 1}p(f_i+1)(1-p(f_i))(1-ρ(f_i,i+1 )) (1) where ρ(f_i,i+1)=ρ(f_i,i+1)=(E[f_i(Sn)f _i+1(Sn)]- p(f_i)p(f_i+1))/(√(p(f _i)-p(f_i)²)(p(f_i+1)- p(f_i+1²))) (2). S number of time slots in a cycle; ρ(f_i,+1) time-slot autocorrelation coefficient; E[x]=expected value of x; p_x=probability of the signal x being “one”. The switching activity analysis of a signal at the word level is computed by summing the activities of all the individual bits constituting the signal. It is also shown that if the correlation coefficient of the higher order bits of a normally distributed signal x is ρ(x_c), then the bit P₀ where the correlation begins and the correlation coefficient is related hy ρ(x_c)=erfc{(2(P₀-1)-1)/(√2σ_x )} where erfc(x)=complementary error function; σ_x=variance of x. The proposed approach can estimate the switching activity in less than a second which is orders of magnitude faster than simulation-based approaches. Simulation results show that the errors using the proposed approach are about 6.1% on an average and that the approach is well suited even for highly correlated speech and music signals     

On Polar Polytopes and the Recovery of Sparse Representations

Suppose we have a signal y which we wish to represent using a linear combination of a number of basis atoms a_i,y=Sigma_ix_ia_i=Ax. The problem of finding the minimum l₀ norm representation for y is a hard problem. The basis pursuit (BP) approach proposes to find the minimum l₁ norm representation instead, which corresponds to a linear program (LP) that can be solved using modern LP techniques, and several recent authors have given conditions for the BP (minimum l₁ norm) and sparse (minimum l₀ norm) representations to be identical. In this paper, we explore this sparse representation problem using the geometry of convex polytopes, as recently introduced into the field by Donoho. By considering the dual LP we find that the so-called polar polytope P* of the centrally symmetric polytope P whose vertices are the atom pairs plusmna_i is particularly helpful in providing us with geometrical insight into optimality conditions given by Fuchs and Tropp for non-unit-norm atom sets. In exploring this geometry, we are able to tighten some of these earlier results, showing for example that a condition due to Fuchs is both necessary and sufficient for l₁-unique-optimality, and there are cases where orthogonal matching pursuit (OMP) can eventually find all l₁-unique-optimal solutions with m nonzeros even if the exact recover condition (ERC) fails for m.