This work analyzes the operation of a dithered quantizer of picture luminance. It is shown that the addition of dither before quantization restores some of the pictorial information which a coarse quantizer would otherwise discard. Two-dimensional ordered dither patterns are described which are considerably more effective for this purpose than random distributions of the same dither samples. The patterns of dither can also be designed so that noise artifacts on the output picture are less visible than with equivalent random dither and so that (contrary to the random case) it is not necessary to subtract the same dither pattern at the receiver for substantially best results.

Extraction of a timing wave from PCM baseband signals through a square law device is studied. The relationship between waveforms and systematic jitter in the timing wave is derived. A criterion for jitter free equalization is obtained. Examples of waveforms that satisfy the criterion are given. Maximum jitters for some practical waveforms are given by numerical examples.

The probability density function and autocorrelation function of random FM in a mobile radio system are derived for the case where postdetection selection diversity is used. Closed-form solutions for the autocorrelation function have been found for no diversity and two-branch diversity and infinite series expansions for higher orders. The use of two-branch selection diversity is shown to reduce the random FM by 13.5 dB in a typical case, with a further 2.5-dB reduction when three branches are used. The effect of other forms of diversity on random FM is briefly assessed, but not pursued in detail.

Consideration is given to the use of the discrete Kalman filter as an equalizer for digital binary transmission in the presence of noise and intersymbol interference. When the channel is modeled as an n -tap transversal filter, the Kalman filter assumes a similar form with feed forward and feedback. It is shown how the Kalman filter can be used to estimate both the tap weights and the binary signal. Computer results on a fixed 6-tap channel show that use of the 6-tap Kalman filter yields a considerably smaller error probability than when a conventional transversal equalizer with 15 taps is used. Limited computer studies on the same channel, assumed to be initially unknown, suggest that the Kalman filter is capable of converging rapidly in the adaptive mode. Though these results are very encouraging, much work remains in the study and optimization of performance in the adaptive mode.

This paper describes an experiment that was performed to study the subjective effects of noise and loss on actual telephone calls made within the Murray Hill location of Bell Telephone Laboratories. The effects of noise and loss on talker volume as well as on transmission quality ratings are discussed. A joint transmission grade of service is obtained, which shows, for various distributions of noise and loss in the circuit, the expected percentage of customers rating the circuit quality as "good or better," and "poor or worse."

The paper describes an experimental system constructed to allow investigation of pseudorandom dither signals for coarse quantization of 625-line monochrome television siginals. The system employs a separate encoder/decoder for the television synchronizing pulses so that the picture signal can be allowed to occupy the full nonsaturating range of the video encoder. Dither samples are added to the video signal before coding and subtracted at the decoder and are negatively correlated to maximize the picture signal-to-noise ratio according to a subjective noise weighting function. The power spectral distribution of the quantizing noise is derived and, assuming this to be picture independent and random, pre- and deemphasis networks are used to increase the signal-to-noise ratio. Empirical methods are used to determine the network parameters. The results of preliminary subjective experiments with the 3 bits/sample pseudorandom codec are given for a sampling frequency of 12 MHz and indicate the practical limitations of a theory which predicts a high standard of picture quality.

An easily implemented estimator of the amplitude (including sign) of a sine wave of known phase is described. The performance of this estimator when zero-mean Gaussian noise is added to the signal is then determined by computing its probability density, mean, and second moment. These statistics are shown to converge to those of the maximum likelihood estimate for large values of signal-to-noise ratio.

Terminal analog or formant speech synthesizers have found many applications in speech research. These include investigation of computer voice response, speech synthesis-by-rule, and speech perception studies, among others. Many types of formant synthesizers have been designed and realized either in analog circuitry or as a computer program. In this paper we describe a digital hardware realization of a formant synthesizer which utilizes the technique of digital multiplexing of a single arithmetic unit among several digital filter sections. The advantages of this hardware over conventional analog hardware include: precise control over center frequencies and bandwidths of the resonators in the synthesizer, stability and reliability of the hardware, light weight, small size, and low power consumption. The synthesizer is capable of producing speech in real time at sampling rates up to 12.8 kHz, using 24 bits to process the digital signals internal to the synthesizer. A 12-bit digital-to-analog convertor supplies an immediate analog output for monitoring the speech and a provision is included for returning 16 bits of the output signal to the computer for future processing such as waveform display or spectrum analysis.