Bounds on maximum throughput for digital communications withfinite-precision and amplitude constraints

The problem of finding the maximum achievable data rate over a linear time-invariant channel is considered under constraints different from those typically assumed. The limiting factor is taken to be the accuracy with which the receiver can measure the channel output. More precisely, the following problem is considered. Given a channel with known impulse response h(t), a transmitter with an output amplitude constraint, and a receiver that can distinguish between two signals only if they are separated in amplitude at some time t ₀ by at least some small positive constant d, what is the maximum number of messages, N_max, that can be transmitted in a given time interval [0,T]? Lower bounds on N_max can be easily computed by constructing a particular set of inputs to the channel. The main result is an upper bound on N_max for arbitrary h(t). The upper bound depends on the spread of h(t), which is the maximum range of values the channel output may take at some time t₀>0 given that the output takes on a particular value α at time t=0. Numerical results are shown for different impulse responses, including two simulated telephone subscriber loop impulse responses     

Revisiting the spread spectrum sliding correlator: why filtering matters

A wireless channel sounder based upon the conventional spread spectrum sliding correlator implementation uses unfiltered pseudo-random noise (PN) at both the transmitter and receiver to generate a time-dilated copy of the channel?s impulse response. However, in addition to this desired impulse response, the sliding correlator also produces a noise-like, wideband distortion signal that decreases the measurement system?s dynamic range. Careful selection of the sliding correlator?s lowpass filter can significantly reduce this distortion, but no amount of filtering will remove it completely. In contrast, using filtered PNs at both the transmitter and receiver enables one to remove this distortion in entirety and realize a measurement system whose dynamic range closely approximates the theoretical ideal for spread spectrum systems.     

On optimal signal sets for digital communications with finiteprecision and amplitude constraints

The maximum data rate that can be reliably communicated given a linear, time-invariant, dispersive channel, a receiver that samples the channel output to within an accuracy of ±d where d >0, and a transmitter with an output amplitude constraint is evaluated. For any dispersive channel the maximum rate depends on d and is finite. The transmitted waveforms must be designed so that two channel outputs associated with two distinct transmitted signals are separated in amplitude at a particular time by d. It is shown that given any channel impulse response with rational Laplace transform, there exists an optimal sets of inputs that are ±A everywhere where A is the maximum allowable amplitude. Furthermore, in any finite time interval, each input changes sign a finite number of times. If the channel impulse response is a single decaying exponential, it is shown that simple binary signaling, in which A or -A, depending on the current message bit, is transmitted during each symbol interval, maximizes the data rate     

A two-dimensional RAKE receiver architecture with an FFT-based matched filtering

This work proposes a two-dimensional (2-D) RAKE receiver, which is a spatial-temporal matched filter implemented in the frequency domain. To form a beam pattern, we calculate the spatial frequency spectra of received signals on the antenna array using fast Fourier transform (FFT). After FFT beamforming, a bank of FFT-based matched filters is used to perform code matching. Afterward, the code-matched signals are summed up with maximal-ratio combining through a spatial-temporal channel-matched filter implemented in the frequency domain. This 2-D RAKE receiver includes a channel sounder that is used to estimate the spatial and temporal channel impulse response parameters, such as delays, directions of arrivals, and complex gains of multipath components. Monte Carlo simulations have been used to evaluate the receiver bit-error rate performance in both static channel and mobile radio channel environments. Simulation results show that the RAKE receiver performs well in both kinds of channels.     

Performance of DS/SSMA communications in impulsive channels. II.Hard-limiting correlation receivers

For pt.I see ibid., vol.COM-35, no.11, p.1179-88 (1987). In part I it was demonstrated that impulsive channel noise can be a serious detriment to the performance of direct-sequence spread-spectrum multiple-access (DS/SSMA) communications when conventional linear correlation reception is used. Here, a hard-limiting correlator as an alternative for reception of multiple-access transmission in impulsive channels is considered. For K asynchronous binary PSK DS/SSMA users sharing a linear channel corrupted by impulsive noise that is modeled at the output of the front-end filter of the receiver, techniques are developed for analyzing bit error probabilities of this hard-limiting receiver by exact computation for short spreading sequences, by approximation for longer spreading sequences, and by asymptotic limits for infinitely long spreading sequences. Performance is compared to that of the linear correlator under a variety of conditions, showing that hard-limiting correlation reception can offer substantial improvement over conventional systems in impulsive channels. However, the linear receiver is more effective against multiple-access noise only, and so a tradeoff emerges between rejection of impulsive noise and rejection of multiple-access interference     

Comparison of DFE and MLSE Receiver Performance on HF Channels

Data communication at rates near or above 2 kbits/s on 3 kHz-baadwidth HF radio channels is subject to impairment from severe linear dispersion, rapid channel time variation, and severe fading. In this investigation, recorded 2.4 kbit/s QPSK signals received from HF channels were processed to extract a time-varying estimate of the channel impulse response. From the estimated channel impulse responses, performance-related parameters were computed for ideal matched filter reception, maximum-likelihood sequence-estimation (MLSE), and decision feedback equalization (DFE). The results indicated that the simpler DFE receiver suffered only a small theoretical performance degradation relative to the more complex MLSE receiver. Other HF channel impulse response statistics were also obtained to shed light on equalization and filter adaptation techniques.     

Multicarrier modulation with blind detection capability using cosine modulated filter banks

The cosine modulated filter bank (CMFB) is introduced as a multicarrier modulation (MCM) technique for wideband data transmission over wireless channels. Under the name discrete wavelet multitone modulation, CMFB has been considered for data transmission over digital subscriber lines. We propose a new receiver structure that is different from those proposed previously. The new structure simplifies the task of channel equalization, by reducing the number of equalizer parameters significantly. We also propose a novel blind equalization algorithm that fits very nicely in the proposed structure. Moreover, we discuss the bandwidth efficiency of the proposed CMFB-MCM system and show that it is superior to the conventional (single carrier) quadrature amplitude modulation (QAM) and orthogonal frequency-division multiplexing (OFDM). The CMFB is found to be a signal processing block that stacks a number of vestigial sideband modulated signals in a number of overlapping subchannels in the most efficient way. The proposed CMFB-MCM is also compared to OFDM with respect to bit-error rate performance. Under the conditions that the channel impulse response duration remains less than the length of cyclic prefix, OFDM is found marginally superior to CMFB-MCM. However, OFDM degrades very fast when the channel impulse response duration exceeds the length of the cyclic prefix. CMFB-MCM, on the other hand, is found less sensitive to variations in channel impulse response duration.     

Bounding the capacity of saturation recording: the Lorentz modeland applications

The authors consider the problem of bounding the information capacity of saturation recording. The superposition channel with additive Gaussian noise is used as a model for recording. This model says that for a saturation input signal, x(t) (i.e., one that can assume only one of two levels), the output can be expressed as y(t)=x˜(t)+z(t ) where x˜(t) is a filtered version of the input x(t) and z(t) is additive Gaussian noise. The channel is described by the impulse response of the channel filter, h(t), and by the autocorrelation function of the noise. A specific example of such a channel is the differentiated Lorentz channel. Certain autocorrelation and spectrum expressions for a general Lorentz channel are derived. Upper and lower bounds on the capacity of saturation recording channels are described. The bounds are explicitly computed for the differentiated Lorentz channel model. Finally, it is indicated how the derived bounds can be applied in practice using physical measurements from a recording channel     

基于均匀信道化滤波器组的研究与设计

信道化作为软件无线电系统和宽带数字接收机的关键技术之一,而多相滤波结构的信道化处理技术和滤波器组的设计一直是研究难点。根据多相滤波结构,本文给出一种有限冲激响应滤波器的设计方法,然后根据均匀信道化的要求设计出均匀滤波器组,设计出的原型滤波器能够无失真采样滤波并能重构。利用等波纹设计的同时,使优化所得通带波纹数量级达到10-3,衰减达到-100dB。利用MATLAB设计及仿真表明,该设计是可靠的,可作为高速率滤波器设计的方法。     

Spectral shaping and digital synthesis of an M-ary timeseries

The use of pulse shaping to control transmitted spectral density precisely is examined. A digital filter architecture is described that not only mitigates the traditional problems of lengthy development intervals and cost manufacturing methods, but offers the additional features of intrinsically coding high-speed binary (M=2) data into M-ary symbols while ensuring highly reproducible, baud-normalized, transmitter pulse shaping. A conceptual basis for the digital synthesis method is first described, including a functional circuit appropriate to the simplest filter realizations. Spectral effects internal to the filter are considered, and a simple method to obtain desired transmitted spectra is outlined. It is shown that even relatively short pulses used in high-level modulation systems lead to impractical memory storage demands; however, the simple expedient of segmenting the finite impulse function greatly reduced the individual memory requirements, though it necessitates intermediate adding operations. Experimental examples illustrate the design methodology for quaternary (M=4) data signals in a Nyquist communication channel and serve as points of reference for addressing performance and design flexibility     

双/多基地综合脉冲孔径地波雷达的信道化接收技术研究

双/多基地综合脉冲孔径地波雷达在发射站为多载频信号同时辐射,在接收站综合形成发射方向图之前需要对各发射信号分量进行分离.尽管采用数字混频和低通滤波器组可以对各发射分量进行分离,但运算量太大,难以实时实现.为此提出采用多相滤波器组信道化接收技术,设计该雷达的多相滤波器组信道化接收机.这种多相滤波器组信道化接收机与低通滤波器组信道化接收相比,运算量减少N²倍(N为信道数).最后给出了该雷达试验系统的实测数据处理结果,验证了这种多相滤波器组信道化接收技术在该雷达信号处理中的有效性.     

A simple derivation of the power spectrum of full-response CPM andsome of its properties

Using a new representation for continuous-phase modulated (CPM) signals, it is shown how a closed-form expression for the power spectral density of full-response M-ary CPM with modulation index J /M can be obtained by straightforward computations. This result is used to provide an explanation of the fact that this power spectral density depends only on J and not on M     

Blind channel estimation via combining autocorrelation and blind phase estimation

Symbol spaced blind channel estimation methods are presented which can essentially use the results of any existing blind equalization method to provide a blind channel estimate of the channel. Blind equalizer's task is reduced to only phase equalization (or identification) as the channel autocorrelation is used to obtain the amplitude response of the channel. Hence, when coupled with simple algorithms such as the constant modulus algorithm (CMA) these methods at baud rate processing provide alternatives to blind channel estimation algorithms that use explicit higher order statistics (HOS) or second-order statistics (subspace) based fractionally-spaced/multichannel algorithms. The proposed methods use finite impulse response (FIR) filter linear receiver equalizer or matched filter receiver based infinite impulse response+FIR linear cascade equalizer configurations to obtain blind channel estimates. It is shown that the utilization of channel autocorrelation information together with blind phase identification of the CMA is very effective to obtain blind channel estimation. The idea of combining estimated channel autocorrelation with blind phase estimation can further be extended to improve the HOS based blind channel estimators in a way that the quality of estimates are improved.     

Multiuser receivers for CDMA systems in Rayleigh fading channels

Multiuser demodulation in relatively fast fading channels is analyzed. The optimal maximum likelihood sequence detection (MLSD) receiver is derived and a general suboptimal receiver to approximate the MLSD is proposed. The performance of the parallel interference cancellation (PIC) and decorrelating receivers is compared. The PIC receiver is demonstrated to achieve better performance in known channels than the decorrelating receiver, but it is observed to be more sensitive to the channel coefficient estimation errors than the decorrelator. At high channel loads the PIC receiver suffers from bit error rate (BER) saturation, whereas the decorrelating receiver does not. The performance of data-aided (DA) and decision-directed (DD) multiuser channel estimation is also compared. DA channel estimation is shown to be more robust than DD channel estimation, which may suffer from BER saturation caused by hangups at high signal-to-noise ratios (SNRs). The impact of channel estimation filter impulse response on the BER is studied by comparing optimal and suboptimal channel estimation filters. The implementation complexity of the decorrelating and PIC receivers is compared in terms of required floating point operations and clock cycles in a practical communication scenario. It is observed that the PIC receiver is only moderately more complex to implement than the conventional matched filter bank receiver, whereas the decorrelating receiver is significantly more complex     

Design of M‐Channel IIR Uniform DFT Filter Banks Using Recursive Digital Filters

In this paper, we propose a method for designing a class of M‐channel, causal, stable, perfect reconstruction, infinite impulse response (IIR), and parallel uniform discrete Fourier transform (DFT) filter banks. It is based on a previously proposed structure by Martinez et al. [1] for IIR digital filter design for sampling rate reduction. The proposed filter bank has a modular structure and is therefore very well suited for VLSI implementation. Moreover, the current structure is more efficient in terms of computational complexity than the most general IIR DFT filter bank, and this results in a reduced computational complexity by more than 50% in both the critically sampled and oversampled cases. In the polyphase oversampled DFT filter bank case, we get flexible stop‐band attenuation, which is also taken care of in the proposed algorithm.     

A stable and distortion-free IIR echo and howling canceler

The authors propose a configuration for an infinite impulse response (IIR) adaptive echo and howling canceller, which consists of a two-channel maximum entropy lattice filter and its inverse filter. The echo is canceled by the adaptive lattice filter, while the signal distortion is eliminated by the inverse lattice. With stability guaranteed without the necessity of testing, the structure costs O (N) multiplications per sampling period. The algorithm can also be greatly simplified for white input cases     

Filtre correcteur de phase pour égaliseurs sous-optimaux

We propose a receiver structure intended for high bit rate transmissions. In order to ensure a good compromise between performance and complexity, the receiver is composed of a transversal phase- corrective filter followed by an equaliser, which takes only partially the channel impulse response into account. We propose a non-iterative technique to adjust the filter tap coefficients, based on the cepstral properties of a minimum phase response. The performance of the filter is evaluated and shown by simulating a transmission system including a multipath mobile radio channel, the filter and a dfse equaliser (decision feedback sequence estimator).     

Suboptimum detection of trellis-coded CPM for transmission onbandwidth- and power-limited channels

The authors consider a communication scheme based on continuous phase modulated (CPM) signals used in conjunction with trellis-coded modulation (TCM) for transmission over a channel affected by Gaussian noise and fading. This scheme provides a power saving due to the coding gain of TCM and a reduced bandwidth occupancy due to the features of CPM signals. Moreover, CPM provides constant-envelope signals. To keep the complexity of the receiver manageable and to be able to use interleaving/deinterleaving techniques to spread the fade bursts, suboptimal detection schemes must be used. Two such schemes, based on coherent and noncoherent detection of CPM, are considered. Their performance is evaluated by computing the computational cutoff rate of the discrete channel generated by CPM     

Multiple differential detection of continuous phase modulationsignals

A sequence estimation algorithm for the differential detection of the continuous phase modulation (CPM) signals, yielding significant gains in BER performance and with considerable resistivity to fading, is introduced. These advantages, along with the reduced hardware complexity, low cost, and fast synchronization which characterize the differential detector, make the proposed receiver useful for land mobile radio and mobile-satellite communications. The new receiver is based on multiple differential detection. The multiple differential detection strategy provides the decoder with more information regarding the transmitted data and applies a noise decorrelation process on the received signal, useful to the sequence estimation. The algorithm is derived in a general form, and can be applied on any CPM scheme, with any degree of complexity. The authors have evaluated the receiver for two of the most popular CPM schemes, the tamed frequency modulation (TFM) and Gaussian minimum-shift keying (GMSK) (with B₁ T=0.25), in the presence of additive white Gaussian noise (AWGN) and Rician fading. The BER performance evaluation results indicated significant gains and considerable reduction of error floors. In AWGN improvements close to 9 dB have been verified     

Mean-Square Error Optimization of Quadrature Receivers for CPM with Modulation Index 1/2

Detection of continuous phase modulation (CPM) signals can be accomplished using quadrature coherent detectors which include a pair of linear and time-invariant postdetection filters. Performance of the quadrature detectors is highly sensitive to the postdetection filters response. This paper presents a rigorous derivation of an optimum postdetection filter response. The derivation is based on minimizing the mean-square interference subject, to the constraint that the filter noise bandwidth is held constant. The amount of computations involved is fairly small and increases linearly with the receiver observation interval. Performance analysis results for several modulation techniques are presented.