A method of sequence detection without the knowledge of the channel response is proposed. The Viterbi algorithm is extended to a blind form by introducing a new branch-metric, defined as the minimum of the short time-average of the squared error (y/spl circ/k-yk)/sup 2/, where yk and y/spl circ/k are the received signal and its replica, respectively. All possible candidate sequences contained in the short time squared error are defined as trellis states, for which the short time squared error is minimized in respect to a variable of the unknown channel response. The proposed implicit blind sequence detection need not keep each variable after the branch metric is calculated. The consistency of the algorithm is justified by proving that a unique sequence is detected in noise free case. The proof is accomplished under condition that the period of the time-average is longer than the channel response. If the additive white Gaussian noise is assumed, the short time squared errors are minimized beyond the desired minimum by the standard Viterbi algorithm using an apriori known channel response. We call this phenomenon over-minimization. The over-minimization is a major reason of the unavoidable error rate degradation in the blind receiver. An objective of this paper is to establish blind sequence detection for digital mobile communication. Provided the channel response can be regarded as time-invariant during the period of the short time-average of the squared error, the blind sequence detection keeps the same performance as in time-invariant case. In order to improve the error rate performance, an algorithm based on the fractional sampling scheme is introduced. Several error rate performances and behaviors of error events are investigated by computer simulation.>

In contrast to the conventional Viterbi algorithm (VA) which generates hard-outputs, an optimum soft-output algorithm (OSA) is derived under the constraint of fixed decision delay for detection of M-ary digital signals in the presence of intersymbol interference and additive white Gaussian noise. The OSA, a new type of the conventional symbol-by-symbol maximum a posteriori probability algorithm, requires only a forward recursion and the number of variables to be stored and recursively updated increases linearly, rather than exponentially, with the decision delay. Then, with little performance degradation, a suboptimum soft-output algorithm (SSA) is obtained by simplifying the OSA. The main computations in the SSA, as in the VA, are the add-compare-select operations. Simulation results of a convolutional-coded communication system are presented that demonstrate the superiority of the OSA and the SSA over the conventional VA when they are used as detectors. When the decision delay of the detectors equals the channel memory, a significant performance improvement is achieved with only a small increase in computational complexity.>

A new technique for parallel transmission of digital and analog signals over a single telephone channel is presented. The modulation scheme treated in this paper is quadrature amplitude modulation, wherein the digital and analog information are mapped separately as in phase and quadrature components. New decision directed schemes which are applicable to parallel transmission are derived by using an error signal measurement defined independent of the analog part. Computer simulation for the parallel transmission scheme of 4.8 kbit/s digital and 2.4 kHz sampled analog sequences is carried out for a typical dispersive telephone channel. Simulation results show that this scheme is useful for high fidelity analog signal transmission.

The development of a new 4800 bit/s data modem, based on real-time digital signal processing by program controlled microprocessors, is described. The modem is fully compatible with the CCITT Recommendations V.27 bis and V.27 ter. In the microprocessor modem, all the digital signals are processed by one general purpose microprocessor unit only and no hardware multiplier is employed. A series of processing cycle reduction tactics have allowed an off-the-shelf microprocessor to handle all the modem functions in a real-time environment. Particularly noteworthy, among these tactics, is the use of a decision feedback equalization algorithm. Numbers of multiplications, which are required for a conventional linear equalization, are shown to be obviated with the DFE algorithm. It is also shown that the DFE facilitates a certain timing phase control method. The microprocessor modem is extremely flexible because of its software controlled nature, and through programming it will provide a variety of additional functions.

A self-recovering equalization algorithm, which is employed in multilevel amplitude-modulated data transmission, is presented. Such a self-recovering equalizer has been required when time-division multiplexed (TDM) voice or picturephone PCM signals must be transmitted over the existing frequency-division multiplexed (FDM) transmission channel. The present self-recovering equalizer is quite simple, as is a conventional binary equalizer. The convergence processes of the present self-recovering equalizer are shown by computer simulation. Some theoretical considerations on this convergence process are also added.

The zero-forcing automatic equalization technique is extended to equalization in quadrature amplitude modulation data transmission and TV transmission systems. A general cost function, whose minimization problem can be applied to the automatic equalization in the above two systems, is first presented. Based on this cost function, the zero-forcing equalizability condition is solved. Under this condition, simple algorithms, namely unit-increment or amplitude-dependent tap adjusting schemes, can be applied.