Adaptive statistical encoding of d.p.c.m.-coded videotelephone signals

Adaptive statistical encoding of d.p.c.m.-coded video-telephone signals is considered. The need for an adaptive statistical encoder to operate on short-term batch statistics is explained and the effect on the bit rate of limiting the number of codes employed by the statistical encoder is reported.     

Improved digital automatic gain control for p.c.m. signals

A digital automatic gain control is described which uses an open-loop instead of a closed-loop configuration. This change has been made to eliminate oscillations in gain which occurred with sinusoidal input signals in the closed-loop system described previously. Results of a computer simulation of the new system are presented.     

New digital automatic gain control for p.c.m. signals

A new design for digital automatic gain control for p.c.m. telephony is presented. This new design is capable of processing 30 speech channels. It avoids putting highest gain where there is no speech. It has manual and automatic facilities.     

High-performance c.c.d. channel filter for a p.c.m. codec

A monolithic c.c.d. filter for p.c.m. codec was fabricated employing both a minimum-phase design with a substantially reduced number of taps and almost perfect elimination of excess capacitances. Results, such as negligible degradation in frequency responses, considerable reduction of common-mode signal, low filter noise (?78 dBm) and wide dynamic range (84 dB) were successfully obtained.     

Effect of nonrandom digital signals on p.c.m. frame synchronisation

It is shown that for certain nonrandom p.c.m. digital signals a suitable choice of frame alignment word can result in an improved system realignment performance.     

Stability of p.c.m.-oscillator synchronisation systems with sampled error signals

Previous analyses of control-system stability have approximated the p.c.m. synchronisation system to a continuous linear model. The digital nature of the equipment indicates that a sampled model is more appropriate, and it is shown that, with this model, the system gains have to be reduced. A comparison between `continuous? and `sampled? stability criteria is made.     

Pitch synchronous 1st-order linear d.p.c.m. system

A system called p.s.f.o.l.d. is described which exploits the correlation between successive pitch periods of a speech signal. This system is a differential one and can employ various types of encoders. We describe a p.s.f.o.l.d. system using a 1st-order d.p.c.m. encoder and show that for a speech utterance this system has a peak signal/noise ratio which is 6 dB larger, and has an increase in dynamic range of 13 dB, compared with a 1st-order d.p.c.m. codec.     

Feasibility of using digital filters to interconnect p.c.m. and f.d.m. transmission systems

A time-division multiplex (t.d.m.) and a frequency-division multiple (f.d.m.) system may be interconnected by means of gates and lowpass filters. The use of digital filters (d.f.s) for this purpose appears attractive when the t.d.m. system uses pulse-code modulation (p.c.m.), because digital-analogue and analogue-digital conversion are unnecessary at the interface between the d.f. and the p.c.m. system. The problems involved in applying d.f.s in this application are discussed. It is concluded that the interconnection of p.c.m. systems with f.d.m. systems using quite low carrier frequencies requires the development of digital circuits operating at several hundred megahertz.     

Meander-channel c.c.d. (m.c.c.d.) recursive filter

A modified input-weighting concept for a c.c.d. recursive filter is proposed; it has been implemented using an m.c.c.d. with a complementary input scheme which provides net weighting coefficients. This improved structure requires neither operational amplifiers nor resistors. The operation and performance of the fabricated 2nd-order bandpass filter are presented.     

Partial correction of transmission errors in d.p.c.m. without recourse to error correction coding

A method for partially correcting transmission errors in a 1st-order d.p.c.m. system, without recourse to channel coding, is described. A simple detection algorithm based on the statistical properties of a sequence of differences between adjacent samples of a modified decoded sequence is used to identify the erroneous samples. Three different correcting algorithms are described. Substantial improvements in the decoded waveform are achieved, and, for a Markov input process, an error rate of 0.04% in the first and second most significant digits, an improvement in s.n.r. of 7dB is achieved.     

Meander channel c.c.d. (m.c.c.d.) transversal filter

A simple and compact m.c.c.d. transversal filter is presented. The device features (a) an input-weight technique to achieve a large output voltage, (b) a new `split-input-gate? weighting to provide accurate filter coefficients, and (c) a simplified electrode structure by using the m.c.c.d. as a summing register.     

Threshold region of f.d.m./f.m. signals

Although many papers have been published concerning the analysis of the threshold region of f.m. signals, they are all incomplete. The present letter introduces a new analysis, in which the carrier is modulated by a Gaussian baseband signal, and gives important results which cannot be deduced from previous analyses.     

Application of c.c.d.s to f.d.m. channel filtering: a phase-cancellation system

A system is suggested for generating and demodulating single-sideband signals, based on the phase-cancellation method; it is particularly attractive for implementation by c.c.d.s, whose well known features it seems fully able to exploit.     

Delta-modulation/p.c.m. convertor

A simple and sufficiently accurate digital convertor is suggested that converts a delta sequence of an exponential delta modulator into a sequence of digital numbers. Each number equals the height of the analogue signal at the input of the delta modulator at each moment. The system employs an up-down counter and some logic based on conventional full adders.     

Simple delay-line p.c.m. coder

A simple p.c.m. coder, using a delay line as the coding network and having signal/noise ratio and other characteristics similar to conventional p.c.m. systems, has been developed. The coder can easily be extended to ternary and multilevel working, and the reconstructed message, available at the transmitter, may be used for purposes of monitoring and further reducing quantising noise.     

Upper bound on signal/quantising-noise ratio for television d.p.c.m. systems

Consideration is given to an upper bound on signal/quantising-noise ratio for television d.p.c.m. systems. The 7.2 dB gain for entropy coding assumes ? entropy of the bit stream of the quantiser output. The calculations are based on Laplacian signals, because television signals at d.p.c.m. quantiser inputs are approximately Laplacian.     

Speech-transmission performance of p.c.m. systems

A method is described for calculating a quantity representative of the transmission performance of a p.c.m. system. The value obtained depends upon (a) the number of quantised output voltage states, (b) the companding law and (c) the volume of the input speech signal relative to the overload point of the system.     

Reducing quantising noise in p.c.m. coders

An analogue-feedback method has been developed to reduce quantising noise in p.c.m. systems. The improvement in signal/noise ratio, however, depends on the loop delay, saturation limit of the coder and the number of digits used. The overall characteristics of the feedback p.c.m. systems have been found to be better than those of the conventional p.c.m. for bit rates up to 50 × 103 bit/s.     

Possible use of pilot signals to indicate delay distortion in operating f.m.-f.d.m. radio systems

Pilot tones normally used in f.m.-f.d.m. transmission allow propagation distortion to be estimated on each path of an operating system. The phases of the 1st-order sidebands relative to the carrier provide a finite-difference approximation to the gradient of group delay with frequency.     

Optimised hybrid unidigit p.c.m. system

The coding efficiency of unidigit p.c.m. systems has been improved by using a secondary feedback loop, and the gain/frequency characteristic of the coder has been equalised. The overall signal/noise characteristic of this hybrid system is found to be better than those of all other unidigit systems; it is inferior to the conventional p.c.m. system only for bit rates higher than 60 × 103 bit/s.