In this paper, a new gradient-based adaptive algorithm for the estimation of discrete Fourier coefficients (DFC) of a noisy sinusoidal signal is proposed based on a summed least mean squared error criterion. This algorithm requires exactly the same number of multiplications as the conventional LMS algorithm, and presents much improved performance in both white and colored noise environments at the expense of some additional memories and additions only. We first analyze the performance of the conventional LMS algorithm in colored additive noise, and point out when its performance deteriorates. Then, a summed least mean squared error criterion is proposed, which leads to the above-mentioned new gradient-based adaptive algorithm. The performance of the proposed algorithm is also analyzed for a single frequency case. Simulation results are provided to support the analytical findings and the superiority of the new algorithm.

This paper presents a variant of the "Third-Order Recursion (TOR)" method for bispectral estimation of transfer-function parameters of a non-minimum-phase all-poles system. The modification is based on the segmentation of system-output data into coupled records, instead of independent records. It consists of considering the available data at the left and the right of each record as not null and taking them as the data corresponding to the preceding and succeeding record respectively. The proposed variant can also be interpreted as a "Constrained Third-Order Mean (CTOM)" method with a new segmentation in overlap records. Simulation results show that this new segmentation procedure gives more precise system parameters than the TOR and CTOM methods, to be obtained. Finally, in order to justify the use of bispectral techniques, the influence of added white and colored Gaussian noise on the parameter estimation is also considered.

To improve the demodulated signal-to-noise ratio, SNR, for colored noise environments, we present a new direct-sequence spread-spectrum receiver system, whose construction is based on the concept of Shaped M-sequence Demodulation (SMD). This receiver has the function for shaping the local dispreading-code waveform. This method can modify the frequency transfer function from a received input to the damp-integrated output according to the power spectrum of colored noise added in the transmission process. SMD performs the combined function of a whitening filter and a matched filter, which can be used to implement an optimum receiver. For the case when the additive colored-noise power spectrum is known and the transmission channel is non-band-limited, a design theory is derived that provides the maximum SNR by choosing the best dispreading-code sequence corresponding to a given signature spreading-code sequence. The noise power component produced in the receiver damp-integrated-output is anayzed by introducing the auto-correlation matrix of the additive noise. The SNR performance of systems, one using non-optimized codes and the other using optimized codes, is examined and compared for various noise models. It is verified by analysis and computer simulation that, compared to a conventional system using non-optimized codes, remarkable SNR improvements can be achieved due to the whitening effect acquired without producing inter-symbol interference. In contrast, if a transversal whitening filter is front-ended, it produces inter-frame interference, degrading the SNR performance. The band-limiting effect of the transmission channel is also analyzed, and we confirmed that the codes optimized for the non-band-limited channel can be applied to the band-limited channel with little degradation of SNR. SMD is inherently tolerant of fast-changing noise such as fading, due to its frame-by-frame operation. Considering this function as a general demodulation scheme, it may be called "Local Code Filtering."

In this paper partial response signalling and trellis coded modulation are considered together to improve bandwidth efficiency and error performance for M-QAM and denoted as Modified/Quadrature Partial Response-Trellis Coded Modulation (M/QPR-TCM) and two new non-catastrophic schemes M/6QPR-TCM and M/9QPR-TCM are introduced for 4QAM. In colored noise with correlation coefficient less than zero, the proposed schemes perform better than in AWGN case. Another interesting result is that when the combined system is used on a Rician fading channel, the bit error probability upper bounds of the proposed systems are better than their counterparts the 4QAM-TCM systems with 2 and 4 states, respectively, for SNR values greater than a threshold, which have the best error performance in the literature.

Until now, in a communication system which deals with multiple processes, system behavior has been described by a fixed number of processes. The state reachability problem for specified processes was generally deliberated within a pre-defined number of processes, and was analyzed by essentially searching for all possible behaviors. However, in a system whose number of processes is arbitrary, a given state which is not reachable in some situations which consists of a small number of processes might be reachable in another situation which consists of a larger number of processes. This article discusses the above problem, assuming that the behavior of a system is described by an arbitrary number of processes. After discussing the relationship between our model and the Petri net model, we clarify the properties between the set of reachable states and the number of processes involved in the system, and show an algorithm to obtain a sufficient number of processes for resolving the reachability problem.

This paper proposes a new adaptive algorithm of the FIR type digital filter for an acoustic echo canceller and similar application fields. Unlike an echo canceller for line, an acoustic echo canceller requires a large number of taps, and it must work appropriately while it is driven by colored input signal. By controlling the filter tap length and updating filter coefficients multiple times during a single sampling interval, the proposed algorithm improves the convergence characteristics of adaptation even if colored input signal is introduced. This algorithm is maned VT-LMS after variable tap length LMS. The results of simulation show the effectiveness of the proposed algorithm not only for white noise but also for colored input signal such as speech. The VT-LMS algorithm has better convergence characteristice with very little extra computational load compared to the conventional algorithm.