Baseband Data Research Articles

Operation of a frequency shift keyed subcarrier multiple-access system for a passive optical network in the presence of strong adjacent channel interference

We use low-pass filtering of baseband data to limit the transmitted RF bandwidth and permit operation of a sub-carrier channel in the presence of two adjacent channels, each with 17 dB higher optical power than the measured channel. Each channel carries 3 Mb/s, and the channel spacing is 6 MHz. Preliminary measurements indicate that an increase in channel spacing to approximately 7.5 MHz will increase the dynamic range beyond 20 dB. >

Convergence analysis of decision-directed adaptive echo cancellers for baseband data transmission

This article deals with the analysis of a decision-directed echo canceller working in a baseband data communication link. This guided scheme, as known in self-adaptive equalization systems, introduces stable local minima in the error surface thus dissuading any gradient search procedure. Regardless of this drawback, the article studies two stochastic gradient algorithms in a multimodal error surface context. They are based in the minimization of absolute and quadratic norms (L/sub 1/ and L/sub 2/) of an improved error reference signal. The analysis assumes bipolar data contaminated with a residual intersymbol interference plus background white noise. The article introduces a new perspective on the analysis of conditions that guarantees convergence towards any desired steady state even with the existence of local minima. It uses a dynamic bounding function for the adaption step dependent on the residual echo variance that allows tracking of the global convergence-condition in the working range. Furthermore, the analysis makes it also possible to predict whether the algorithm will converge or whether it could be trapped in any stable stationary point. It also shows the risky dependence of the convergence towards an undesired steady state on some internal and working variables, such as initial filter settings, interference level, etc.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Use of subcarrier multiplexed acknowledgment tones for contention recovery in WDMA networks

Subcarrier multiplexed acknowledgment tones are used for contention recovery in connectionless WDMA networks with baseband data packets and subcarrier multiplexed control headers. By using this technique, an access node is allowed to retransmit a message (old or new) immediately after contention occurs, effectively increasing the data offered load, and the throughput of the networks is significantly improved.

Performance of WDMA networks with baseband data packets and subcarrier multiplexed control channels

The use of subcarrier multiplexed control channels for wavelength-division multiple access (WDMA) networks transporting baseband data packets is investigated. The contention problem is analyzed, and general expressions are derived for the throughput of slotted and random-arrival networks with N access nodes and Q subcarriers. The throughput of the networks is significantly increased by using multiple subcarriers for control channels. >

Performance of multiple access WDM networks with subcarrier multiplexed control channels

In this paper, the performance of multiple access WDM networks with baseband data packets and subcarrier mul- tiplexed control channels is evaluated. The advantages of using subcarriers for control channels are described. The contention problem is analyzed and general expressions are derived for the throughput of slotted and random-arrival networks with M access nodes and Q subcarriers. It is shown that using multiple subcarriers for control channels can significantly improve the throughput of the networks, especially when the traffic is heavy or the data packets are short.

Five Gb/s operation of a 50-channel optical interconnect

The architecture and experimental demonstration of a novel optical time-division multiple-access (TDMA) interconnect is presented in detail. Optical multiplexing and synchronization is used to overcome the electronic multiple-access bottlenecks associated with gigahertz-bandwidth multiprocessor communication systems. A self-clocking optical TDMA interconnect is described that may be more practical to implement than other shared-medium multiple-access protocols, such as frequency division or code division. An experimental optical TDMA interconnect is reported that uses a 100-MHz repetition rate, mode-locked laser with external modulators to generate the base-band data, and operates at a multiplexed data rate of 5 Gbits/s accommodating up to 50 channels. System measurements reveal bit error rates of less than 10 -9 , low channel crosstalk, and subnanosecond multiaccess capability. A power budget analysis predicts that 100 Gbit/s systems with 1000 nodes are feasible.

A low-frequency digital spectrum analyzer for laboratory instruction

The design and implementation of a low-frequency spectrum analyzer is described. The design features a low-cost spectrum analyzer for use in the electrical engineering senior year communications laboratory. The frequency range of the spectrum analyzer extends from under 1 Hz up to 20 kHz. It covers the spectral region of interest in speech processing and baseband data transmission, and provides the students with a flexible signal analysis tool.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Fast On-Line Speech/Voiceband--Data Discrimination for Statistical Multiplexing of Data with Telephone Conversations

This paper concerns the design of a speech/data discriminator with the speed and accuracy required for statistical multiplexing of speech and high-speed baseband data on a single telephone line. The structure proposed for the discriminator combines the information from two parallel processes. The first process, even though slow, accurately recognizes the speech/data nature of the signal. This nature is determined by statistical pattern classification applied to simple zero-crossing-type parameters extracted from the signal filtered in a PCM representation. The second process has temporal accuracy in detecting transition events, from speech to data and vice versa. This detection is provided by special parameters which supply tentative transition markers. This paper applies these processes to the case of 9600 bit/s modems and describes a complete discrimination algorithm which is fit for microprocessor technology. The simulation results provided indicate a very low error rate and demonstrate that transitions are detected exactly or within one or two sampling intervals.

A dynamic programming algorithm for simultaneous phase estimation and data decoding on random-phase channels

The problem of simultaneously estimating phase and decoding data symbols from baseband data is posed. The phase sequence is assumed to be a random sequence on the circle, and the symbols are assumed to be equally likely symbols transmitted over a perfectly equalized channel. A dynamic programming algorithm (Viterbi algorithm) is derived for decoding a maximum {\em a posteriori} (MAP) phase-symbol sequence on a finite dimensional phase-symbol trellis. A new and interesting principle of Optimality for simultaneously estimating phase and decoding phase-amplitude coded symbols leads to an efficient two-step decoding procedure for decoding phase-symbol sequences. Simulation results for binary, <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">8</tex> -ary phase shift keyed (PSK), and 16-quadrature amplitude shift keyed (QASK) symbol sets transmitted over random walk and sinusoidal jitter channels are presented and compared with results one may obtain with a decision-directed algorithm or with the binary Viterbi algorithm introduced by Ungerboeck. When phase fluctuations are severe and when occasional large phase fluctuations exist, MAP phase-symbol sequence decoding on circles is superior to Ungerboeck's technique, which in turn is superior to decision-directed techniques.

Useful Signaling Waveforms and Related Transmit Filter Functions in Bandwidth Limited Channels

A collection of easily generated signaling waveforms for a binary baseband data system is presented along with the required transmit filter functions. Filtering is based on the criterion of zero intersymbol interference and optimum P <inf xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">e</inf> versus receiver input signal-to-noise ratio. The channel considered is a linear one of bandwidth equal to <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">[(1 + \alpha)/2]</tex> Hz/baud; α being allowed to vary between zero and one. The results apply, as well, to multilevel data and to bandpass systems, such as digital radio.

Bit-level synchronisation in microprocessor networks

A synchronisation method for baseband data transmission in microprocessor networks is developed. The method is applicable in systems using ADCCP-, HDLC- or SDLC-protocols and NRZI-code and point-to-point or loop-mode operation. The method is based on bit resynchronisation using counters. No phase-locked loops are used. This provides a very efficient and fast correction response with relatively simple circuitry. The performance of the system is analysed and tested in practice.

High-Resolution Frequency Measurement by Linear Prediction

The efficacy of linear prediction spectrum analysis (LPSA) (also called autoregressive or maximum entrophy spectrum analysis) for problems concerning narrowband signals in noise is examined by means of Monte Carlo simulation. For the case of pure tones (sinusoids) in white noise, both frequency estimation accuracy and resolution of two closely spaced tones with the linear prediction approach are discussed. In addition, estimation of the carrier frequency of a narrowband random (fading) Gaussian signal in noise is considered. The results are compared with standard fast Fourier transform (FFT) methods. These comparisons indicate that there is no significant advantage to using the linear prediction approach with real (bandpass) data. It is concluded that these experiments should be repeated with complex (baseband) data before a complete assessment of the LPS approach can be made.

Digital echo cancellation for baseband data transmission

In analogy with analog speech transmission, simultaneous two-way ("full-duplex") transmission of data signals over two-wire circuits can, in principle, be achieved by the application of hybrid couplers. In practice the resulting imperfect isolation between transmitter and receiver at the same end of a connection is disastrous for the detection of the data from the other end and has to be nullified in some way. A solution to this problem can be found in echo cancellation by means of an adaptive digital filter. In this paper an integrated circuit realized in 4Φ-dynamic NMOS technology is presented which has been designed especially for echo cancellation in baseband modems. In the design a number of aspects have extensively been accounted for. These are the choice of adaptation algorithm, the interaction between the digital filter and its analog environment, interpolation, finite precision implementation, and the natural occurrence of an effect akin to dithering. These subjects receive full attention in this paper. The realized IC can be used for the economical upgrading of existing baseband modems to full-duplex service and can pave the way to the use of similar circuits in the local area of the future digital telephone.

Baseband compatible s.a.w. processors

A nonlinear acoustic tapped delay line is shown to be capable of providing signal processing at baseband data format via a classical amplitude-modulation/demodulation process; compatibility with baseband-operating analogue/digital arithmetic units is simply and directly achieved, thus resulting in a potentially attractive alternative to existing techniques for programmable transversal filtering implementation.

Some Studies on Partial Response Signalling Through Voice-Grade Circuits

With the current practice of using FSK, DPSK and YSB for data transmission, only 60% of the available bandwidth of a long-distance voice-grade circuit (300–3 400 Hz) is utilised. On the other hand, PRS codes have the promise of transmitting data at the ideal rate of 2 bits/Hz. Theoretical and experimental studies on some useful PRS codes have been made to evaluate their performances for baseband data transmission through voice-grade circuits, it has been shown here that by the use of suitable equalisers—compromise as well as adaptive, the 3-level PRS code is most suitable for signalling rates upto 7.2 kbps through voice-grade circuits without any spectrum translation. The eye pattern speed/phase tolerances and error rates have been evaluated and found to be satisfactory at data rates of 6.4 kbps and 7.2 kbps. For band-pass channels with higher highpass cutoff, higher-order PRS codes are more suitable and their performances have been found to be satisfactory as well.

Probability of error in digital fiber optic systems with inter-symbol interference and signal dependent additive noise

Data transmission via optical fiber is a new discipline of communication theory. The principal difference from conventional baseband data transmission, which is characterized by a signal independent additive Gaussian noise, is the existence of a signal dependent shot noise. This paper presents a technique for estimating the error probability performance of digital systems with inter-symbol interference and signal dependent additive noise. For binary antipodal (±1) systems, the approximate upper bound to the error probability is twice the lower bound. Hence either can be taken as a good approximation to the actual error probability. The technique is then applied to a model of some promising optical data communication systems and a good approximation to the error probability is obtained. Some observations about the effect of various system parameters on the error probability and some numerical examples are presented.

A digitally controlled adaptive equalizer for baseband data communication

An adaptive equalization method for high-speed and/or long-haul baseband data transmission is described which uses the pulse length as a control criterion. The method does not depend on the attenuation of the signal and can be applied to all types of switched cables as well as overhead lines.

Spectrum Shaping with Alphabetic Codes with Finite Autocorrelation Sequence

In this paper we consider the problem of recoding binary data into ternary sequences for synchronous baseband data transmission systems. Our major objective is the shaping of the power spectral density of the channel process in order to combat the effects of intersymbol interference caused by the low-frequency cut-off of the channel. This motivates our interest in designing codes whose channel process has small Power in the low frequency range. Based on the MacLaurin expansion of the power density function, we use as a design heuristic the normalized second derivative Δ of the latter. To make computationally manageable the expression of Δ, we restrict ourselves to a class of codes whose autocorrelation sequence contains a finite number of nonzero terms, and present a corresponding code design algorithm. As examples, we exhibit two new codes which in many respects improve over the spectra of corresponding well-known codes.

An improved receiver for baseband data communication using a digitally controlled AGC

A suitable Automatic Gain Control (AGC) in baseband data communication receivers is essential for a low error rate of the link. Analogue AGC suffers from conflicting demands of time constants. This paper describes a receiver using digitally controlled AGC overcoming these difficulties.

The Jitter Performance of Phase-Locked Loops Extracting Timing From Baseband Data Waveforms

Phase comparators used in phase-locked loops extracting symbol timing from baseband data waveforms typically only produce a useful error signal when a data transition occurs. This gating of the error signal by data transitions makes the natural model for studying jitter in such phase-locked loops time-varying and difficult to analyze. In this paper we show how, under good approximation, it can be simplified to a time-invariant model that is easily analyzed. Using this model, we study jitter accumulation along chains of digital repeaters with phase-locked-loop timing extractors. A numerical example is given.