Cellular radio started in the early 1980s by using analog technologies. Research in voice coding, modulation, and channel coding resulted in second-generation cellular radio based on digital technologies, which were introduced in the early 1990s. These were all based on advanced time division multiple access technology, resulting in better capacity and lower cost. Today, these digital technologies-based on the Global System for Mobility, digital Advanced Mobile Phone System, and personal digital cellular-have more than 100 million subscribers worldwide. The next step is to introduce wide-band packet services for wireless Internet up to 2 Mb/s. These so-called third-generation systems (Universal Mobile Telecommunications Services, IMT2000) are planned to be introduced in the early 2000s.

The aim of this study is to achieve a simple speech codec at low bit rate for mobile radio applications. The vocoder part is obtained by using a low order Discrete Fourier Transform instead of a filter bank. Different baseband waveform coding schemes combined with different baseband bandwidths have been investigated. In particular we have studied a simple scheme for time encoded baseband in which the time intervals between zero crossings are quantized and transmitted. Amplitude information is transmitted with one bit for every zero crossing. The results concerning speech quality and implementation indicate that a simple 4.8 kb/s codec is feasible. It can be implemented with a few low-speed, low-power CMOS LSI circuits since no multiplications are required.

A study of the asymptotic performance of delta coding with delayed decision is presented for the cases of first and second order integration assuming a stationary, bandlimited Gaussian source. The resulting signal-to-distortion-ratio serves as an upper bound for any delayed (multipath) decision scheme provided that the sampling rate exceeds the Nyquist rate by a factor greater than about two. The analysis shows that the improvement offered by delayed encoding relative to ordinary delta modulation has its basic origin in diminished overload distortion. Some insight is given into how the improvement is related to the decoding filter. Finally, numerical results are given for some sources with speech-like spectra

This concise paper is concerned with the problem of improved delta-coding by using delayed decision instead of bit-bit-by-bit decision. It is found that delayed encoding allows a fairly general predictor to be used without causing instability problems. Simulations indicate that considerable improvement can be achieved by matching the feedback filter to the input process. Delayed encoding requires a search algorithm for making the decisions. Some proposals of algorithms that are efficient from a computational point of view are presented. Particular interest is attached to a highly truncated version of the Viterbi algorithm, which seems very promising.

A class of adaptive schemes for delta modulation is considered whereby the decision about step size is a function of the present and the lastksymbols. Such schemes are studied by computer simulations using speech-like signals through evaluation of SNR. Occasional large overshoots appear in the reconstructed signal after the integrater, and to avoid such events it is proposed to store a few signal samples and decide on the proper bit stream taking into account the adaptive logic. Two quadratic criteria have been tried, and improvements of a few decibels have been established with six stored signal samples and a sampling rate of 48 and 24 kHz. Results indicate that other criteria functions may achieve still higher values of SNR.