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Communication Technology, IEEE Transactions on

Issue 6  Part 1 • Date December 1971

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  • [Front cover and table of contents]

    Page(s): 0
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    Freely Available from IEEE
  • Oracular Vision

    Page(s): 870 - 871
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    First Page of the Article
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  • A 36-Mbit/s Television Codec Employing Pseudorandom Quantization

    Page(s): 872 - 879
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    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. View full abstract»

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  • The Effect of Dither on Luminance Quantization of Pictures

    Page(s): 879 - 888
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    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. View full abstract»

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  • Adaptive Variable-Length Coding for Efficient Compression of Spacecraft Television Data

    Page(s): 889 - 897
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    An adaptive variable length coding system is presented. Although developed primarily for the proposed Grand Tour missions, many features of this system clearly indicate a much wider applicability. Using sample to sample prediction, the coding system produces output rates within 0.25 bit/picture element (pixel) of the onedimensional difference entropy for entropy values ranging from 0 to 8 bit/pixel. This is accomplished without the necessity of storing any code words. Performance improvements of 0.5 bit/pixel can be simply achieved by utilizing previous line correlation. A Basic Compressor, using concatenated codes, adapts to rapid changes in source statistics by automatically selecting one of three codes to use for each block of 21 pixels. The system adapts to less frequent, but more dramatic, changes in source statistics by adjusting the mode in which the Basic Compressor operates on a line-to-line basis. Furthermore, the compression system is independent of the quantization requirements of the pulse-code modulation system. View full abstract»

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  • Delayed Encoding: Stabilizer for Adaptive Coders

    Page(s): 898 - 907
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    The decision process in source encoders can be influenced favorably by anticipating future quantizing errors and modifying the quantizer appropriately. This requires that the signal code be delayed slightly from the corresponding input signal sample. As an adaptation of an existing coder, little advantage is obtained [8]. However, the process has a stabilizing influence so that much stronger adaptation algorithms can be used to advantage, increasing the signal to quantizing noise ratio(S/N)markedly. It is believed that this fact is of general applicability, but it is shown herein only for 1-bit coders. A family of 1-bit coders (delta modulators) using exponentially adaptive step size, with two steps of integration in the feedback path has been studied using a special purpose computer facility. Such coders are ordinarily unstable and useless, but with error anticipation a measuredS/Nadvantage of several dB over optimized adaptive coders of previous design is obtained. The study has concentrated on picture signals and an encoding which does not require a separate channel or code to signal changes in the coder. Fig. 8 compares the optimized delayed encoder operation with an optimized adaptive coder without delay. "Optimization" in the former case requires a modification of the feedback network-the use of two steps of integration instead of one. Delayed encoding is not just an improvement for existing differential coders, it promises to be a revolution in coder design. View full abstract»

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  • A Differential Pulse-Code-Modulation Codec for Videotelephony Using Four Bits Per Sample

    Page(s): 907 - 912
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    A video codec is described that encodes a 1-MHz analog Picturephone® signal into a 6.312-Mbit/s digital form, suitable for transmission on the Bell System T2 digital facility. Through efficient coding of the horizontal sync format, the bulk of the horizontal blanking period is available for sending active video information. The signal is differentially encoded into 4 bits per sample with a quantizer whose levels are spaced nonlinearly. Excess bits are stored in a small buffer to be transmitted during the time made available in the blanking region. The nonlinear quantizer characteristics are subjectively matched to visual processes to minimize the viewer's sensitivity to coding inaccuracies for a broad range of pictures. All the necessary automatic maintenance and alarm functions required for a digitally switched network are included in the codec design. View full abstract»

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  • Transient-Mode Buffer Stores for Nonuniform Code TV

    Page(s): 913 - 922
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    Buffer stores are needed at both sender and receiver whenever a real-time signal is coded into a varying rate sequence of digits and these are transmitted over a channel at a uniform rate. Practical sending stores must be finite in size and therefore are subject to overflow; they thus function in a transient mode. In this paper, deterministic constraints on the operation of sending and receiving stores are established and their sizes are related to bittransmission rate and storage delay. The random behavior of the sending store is modeled as a first-order Markov chain with an absorbing state (overflow). The study is motivated by the case of nonuniformly coded differential PCM television signals. Some simulator measurements are reported on the actual incidence of overflows with such signals when using small-capacity stores (<120-bit storage). The Markov model for these cases is seen to give correct trends, although overall the measurements reveal considerably greater incidences of overflow. It can be expected, however, that the agreement between model and measurement would improve with size of stores. View full abstract»

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  • Variable-Length Redundancy Removal Coders for Differentially Coded Video Telephone Signals

    Page(s): 923 - 926
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    A study of the statistics of a differential pulse-code modulation coder for Picturephone® signals which uses a three-bit uniform length code suggests that a higher transmission efficiency can be obtained with a variable word length code. This permits an increase in number of quantizing levels used, thus improving picture quality. A quantizer with 24 levels rather than eight can be used provided that a buffer store of 10 000 bits is included in the system to smooth the irregular data flow caused by the varying code word length. View full abstract»

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  • The Effect of Channel Errors in the Differential Pulse-Code-Modulation Transmission of Sampled Imagery

    Page(s): 926 - 933
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    This paper presents an analysis, simulation, and discussion of the effects of communication errors on four-bit differential pulse-code modulation (DPCM) sampled imagery. Simulations are presented that describe the effects of inserting periodic "PCM updates" in order to correct communication errors in the DPCM transmission of photographic scenes that have been scanned and sampled at the Nyquist rate. View full abstract»

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  • An Adaptive Dual-Mode Coder/Decoder for Television Signals

    Page(s): 933 - 944
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    An adaptive dual-mode coding system for television signals is described. Its main features are a low bit rate (1.5 bits per sample), the high quality of the reproduced picture, and its moderate hardware. The system is based on the statistical properties of video signals. Specifically, it makes use of the nonuniform spectrum of video signals in the form of a differential scheme containing linear prediction. Furthermore, areas of small amplitude changes between consecutive samples whose probability of occurrence is high are encoded with a reduced coding alphabet. Transients representing sharp edges in the picture are encoded and reproduced with little slope overload and busyness. A method for the buffering of the asynchronous data stream produced by the coder to match a synchronous channel is given. View full abstract»

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  • An Edge-Adaptive Three-Bit Ten-Level Differential PCM Coder for Television

    Page(s): 944 - 947
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    A modified 3-bit per sample differential PCM coder that is subjectively comparable to a 10-level differential PCM coder is described. The accuracy of coding the signals preceding and following large signal changes is restricted. The bits thus saved are used to improve the coding accuracy of large signal changes. No visible impairments are introduced by this process. The changes in coding strategy are carried in the main signal stream to the receiver and do not require flags or buffer memory. The coder's algorithm, implementation, and subjective results are described. View full abstract»

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  • Comparison of nth-Order DPCM Encoder With Linear Transformations and Block Quantization Techniques

    Page(s): 948 - 956
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    Two important classes of coding schemes that use the spatial correlation of picture elements in reducing data redundancy are the differential pulse-code modulation (DPCM) and the unitary transform coding techniques. We will study the performance of annth order DPCM system fornranging from 1 to 22 and compare it to the performance of the unitary-transform techniques (Hadamard, Fourier, and KarhunenLoève) in coding two monochrome still pictures. We will also consider the sensitivities of the coding systems to picture-to-picture variations. View full abstract»

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  • Image Coding by Adaptive Block Quantization

    Page(s): 957 - 972
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    A new source encoder called the adaptive block quantizer is proposed for coding data sources that emit a sequence of correlated real numbers with known first- and second-order statistics, Blocks of source output symbols are first classified and then block quantized in a manner that depends on their classification. The system is optimized relative to both the mean square error and the subjective quality of the reconstructed data for a certain class of pictorial data, and the resulting system performance demonstrated. Some interesting relationships between mean square error and subjective picture quality are presented. View full abstract»

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  • Frequency Interleaved Sampling of a Color Television Signal

    Page(s): 972 - 979
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    In the transmission of color television signals over a digital communications system, the analog TV signal must first be converted to digital form. This paper treats the first step in this conversion, sampling of the analog waveform. A significant reduction in sampling rate can be achieved by using several properties of the television signal. The spectrum of the TV signal has the energy concentrated at harmonics of the line and field rates. By choosing a sampling rate that is less than twice the bandwidth and a frequency that is an odd multiple of one-half the line rate, the aliased energy resulting from the sampling process falls in the gaps of the video signal energy. A comb filter is then used to remove most of the aliased energy. The resulting signal shows no appreciable deterioration. The subjective effects of noise added between harmonics of the line rate are discussed. These results are used to determine the requirements of the comb filter. The key parameters of one form of a comb filter are developed. A quantitative evaluation of the comb filter parameters shows that adequate suppression of the aliased energy can be achieved. View full abstract»

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  • Spatial Transform Coding of Color Images

    Page(s): 980 - 992
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    During the past few years several monochromeimage transform-coding systems have been developed. In these systems, a quantized and coded version of a spatial unitary transform of an image is transmitted over a channel, rather than an image itself. In this paper the transform-coding concept has been applied to the coding of color images represented by three primary color planes of data. The principles of spatial transform coding are reviewed and the merits of various methods of color-image representation are discussed. A performance analysis is presented for the color-image transform-coding system. Results of a computer simulation of the coding system are also given. It is shown that, by transform coding, the chrominance content of a color image can be coded with an average of 1.0 bits per element or less without serious degradation. If luminance coding is also employed, the average rate reduces to about 2.0 bits per element or less. View full abstract»

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  • Digital Coding of Color Picturephone Signals by Element-Differential Quantization

    Page(s): 992 - 1006
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    An element differential quantizer has been used to encode three baseband components of a color Picturephone® type signal in real time. The first part of this paper is concerned with an investigation of the extra bits required to transmit chrominance information over that required for a high-quality luminance signal (coded to 4 bits per picture element). We have found that a total allocation of 1 bit per picture element to the chrominance signals leads to a high-quality color display. The second part of the paper concerns an investigation of a more efficient coding format for color sigaals. In this regard, we have determined that the color signal can be packaged into a 6.3-Mbit/s rate by allocating 12 levels for the luminance component, 6 levels for one chrominance component, and 4 levels for the other chrominance component. Only one chrominance component is transmitted each line and the missing component is obtained by line averaging. The best results were obtained by coding chrominance signals that were matrixed such that their color axes lay between theIandQand color difference axes. A scheme is suggested for simply combining this type of coder with an analog color signal that has the chrominance information compressed into the blanking portion of the signal. View full abstract»

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  • Focal Points in Speech Communication Research

    Page(s): 1006 - 1015
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    Advances in several areas of speech communication research are discussed. Areas selected for comment include: 1) digital encoding and transmission of speech; 2) computer synthesis of speech; 3) automatic speaker verification; 4) speech production; 5) speech perception; and 6) interactive design of digital filters. Advances in these areas have been facilitated by two important tools: a) the growing theory of sampled-data systems and b) the general availability of modest-size, fast digital computers. For each area, specific research examples are chosen to represent current activity. Also, the examples are chosen to emphasize the advantages of digital formulation of the problems and to show how one particular laboratory computer facility is used for interactive studies in speech communication. View full abstract»

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  • A Hardware Realization of a Digital Formant Speech Synthesizer

    Page(s): 1016 - 1020
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    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. View full abstract»

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  • An Experimental 9600-bits/s Voice Digitizer Employing Adaptive Prediction

    Page(s): 1021 - 1032
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    An experimental model of a coder for transmission of speech over a 9600-bits/s digital channel was built to demonstrate feasibility of an adaptive prediction-coding technique. After analog-to-digital conversion of the speech input, the coder employs digital processing using a computer type organization. Resonances in the short-term speech spectrum are removed by a nonrecursive digital transmit filter and the resulting uncorrelated signal is coded by an 8000-bits/s direct feedback delta coder. The transmit filter parameters are adapted to the input spectrum by a least squares algorithm involving calculation of short term correlation coefficients of the sequence of input samples. These filter parameters are multiplexed with the delta coder output for transmission to the receiver. A recursive receive filter restores the original speech spectrum. A computer simulation of the voice digitizer was performed to determine the order of the digital filters and to optimize other parameters prior to the design of the experimental model. The results of the simulation and design considerations for the experimental model are described. View full abstract»

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  • A Variable-Step-Size Robust Delta Modulator

    Page(s): 1033 - 1044
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    An optimum adaptive delta modulator-demodulator configuration is derived. This device utilizes two past samples to obtain a step size which minimizes the mean square error for a Markov Gaussian source. The optimum system is compared using computer simulations with the linear delta modulator and an enhanced Abate delta modulator. In addition the performance is compared to the rate distortion bound for a Markov source. It is shown that the optimum delta modulator is neither quantization nor slope-overload limited. The highly nonlinear equations obtained for the optimum transmitter and receiver are approximated by piecewise-linear equations in order to obtain system equations which can be transformed into hardware. The derivation of the experimental system is presented. The experimental "optimum" system, an enhanced version of the Abate delta modulator and a linear delta modulator were tested and compared using sinusoidal, square-wave, and pseudorandom binary sequence inputs. The results show that the output signal-to-noise (SNR) ratio is approximately independent of the input signal power and is subject only to the limitations of the hardware employed. In addition, voice was recorded using these systems. The demodulated voice indicates negligible degradation is caused by the optimum system and by the enhanced Abate system while the linear delta modulator suffers significant degradation at a sampling frequency of 56 k/s. The systems were also tested at 19.2 k/s. At this bit rate, speech recognition, using the experimental "optimum" system, remained completely intelligible. View full abstract»

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  • A Comparison of Orthogonal Transformations for Digital Speech Processing

    Page(s): 1045 - 1050
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    Discrete forms of the Fourier, Hadamard, and Karhunen-Loève transforms are examined for their capacity to reduce the bit rate necessary to transmit speech signals. To rate their effectiveness in accomplishing this goal the quantizing error (or noise) resulting for each transformation method at various bit rates is computed and compared with that for conventional companded PCM processing. Based on this comparison, it is found that Karhunen-Loève provides a reduction in bit rate of 13.5 kbits/s, Fourier 10 kbits/s, and Hadamard 7.5 kbits/s as compared with the bit rate required for companded PCM. These bit-rate reductions are shown to be somewhat independent of the transmission bit rate. View full abstract»

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  • Systems Analysis of a TDM-FDM Translator/Digital A-Type Channel Bank

    Page(s): 1050 - 1059
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    In keeping with the trend to greater use of digital circuits for signal processing, a project was undertaken to realize in an exploratory way an important telecommunication function using as great a proportion of digital hardware as possible. The function chosen is that of theA-channel bank; viz., the frequency division multiplexing (FDM) of 12 voiceband signals onto a single wire. Because of the nature of its operation the device to be described can also perform a translation between FDM analog signals and time division multiplexed (TDM) digital signals. This paper describes the overall system design of the device with particular emphasis on a noise analysis. The principal sources of noise are the A/D conversion points and the roundoff points that occur at the outputs of multipliers. Each noise source is examined in turn and its contribution to the total noise assessed. It is concluded that the A/D conversion points are the most important noise sources and the most costly to deal with. View full abstract»

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  • Partitioning of Digital Filters for Integrated-Circuit Realization

    Page(s): 1059 - 1063
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    The details of the partitioning of a digital filter for integrated-circuit realization are discussed. It is shown that the fourth-order filtering requirement of an all digital channel bank serving 24 channels may be implemented with 40 integrated-circuit chips. View full abstract»

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  • Pulse-Code Modulation to Voice Conversion--Binary Rate Multiplier Differential Pulse-Code-Modulation Decoder

    Page(s): 1064 - 1069
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    An experimental and computer investigation of a 9-bit, 16-MHz binary rate multiplier (BRM) used in converting digital signals differential pulse-code modulation (DPCM) to analog signals (voice) is reported in this paper. The physical electronic circuitry was provided with a stream of real-time digital samples at a 32-kHz rate from a magnetic tape which had been generated by a computer simulation. The output power spectrum of the electronic circuit was observed and compared to that of the computer calculation. Before exciting the experimental circuit with the digital samples the circuit noise in absence of signal was measured at - 11 decibels above a reference noise using a C message weighting filter (dBrnC0). Three different input signals were used to probe the BRM. Two consisted of single sine waves at various levels and in one test ten arbitrary phase superimposed sine waves were used to simulate white noise. Experimental and theoretical results are in good agreement and together demonstrate that the BRM approach is a viable technique for digital-to-analog (D/A) conversion. View full abstract»

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Aims & Scope

This Transactions ceased publication in 1971. The current retitled publication is  IEEE Transactions on Communications.

Full Aims & Scope