Showing papers on "Bandwidth (signal processing) published in 1979"

The Vivaldi Aerial

Abstract: The Vivaldi Aerial is a new member of the class of aperiodic continuously scaled antenna structures and, as such, it has theoretically unlimited instantaneous frequency bandwidth. This aerial has significant gain and linear polarisation and can be made to conform to a constant gain vs. frequency performance. One such design has been made with approximately 10 dBI gain and ?20 dB sidelobe level over an instantaneous frequency bandwidth extending from below 2 GHz to above 40 GHz.

Wireless in-house data communication via diffuse infrared radiation

Abstract: A novel wireless broadcast/multi-access channel is described for flexibly interconnecting a cluster of data terminals located within the same room. The transmission medium is diffusively scattered infrared radiation at 950-nm wavelength. Transmission is low-to-medium speed and the range up to 50 m. Theoretical analysis indicates that the time dispersion limits the transmission bandwidth of the system to 260 Mbit ċ m/s, but background noise produced by ambient daylight reduces the transmission speed below 1 Mbit/s. The transmission properties of the diffuse optical channel are analyzed, and experimental digital links for baseband PCM at 125 kbit/s and PSK 64 kbit/s are demonstrated.

A new selective fading model: Application to propagation data

Abstract: Channel transmission models for use in estimating the performance of radio systems on line-of-sight paths at 6 GHz are explored. The basis for this study is the simple three-ray multipath fade, which provides a channel transfer function of the form H(ω) = a[1 − b exp −j(ω − ω 0 )τ], where a is the scale parameter, b is a shape parameter, τ is the delay difference in the channel, and ω 0 is the (radian) frequency of the fade minimum. This model is indistinguishable from an ideal channel model, within the accuracy of existing measurements. The propagation data that confirm the model were obtained in summer 1977 from a 26.4-mile hop near Atlanta, Georgia. The received power at 24 sample frequencies spaced at 1.1 MHz and centered on 6034.2 MHz was continuously monitored and recorded during periods of anomalous behavior. The model is applied to estimating the statistics of the channel delay difference, τ. The average delay difference giving rise to significant selectivity in the channel is between 5 and 9 ns. The distribution of delay difference is obtained for delay differences greater than 10 ns. The channel is found to have more than 3 dB of selectivity (difference between maximum and minimum attenuation in band) due to delay differences greater than 20 ns for more than 70 seconds in a heavy fading month. (This is comparable to the time the channel attenuation of a single frequency exceeds 40 dB.) The three-path model requires further simplification for narrowband channel application. For a channel with 30 MHz bandwidth, a model with fixed delay of 6.3 ns provides a sufficiently accurate representation of all observed channel conditions. The resulting nonphysical model is used to statistically characterize the condition of the fading channel. The statistics of the parameters of the fixed delay model are almost independent and of relatively simple form. The distribution of the shape parameter b is of the form (1 − b)2.3. The distribution of a is lognormal.

Spread spectrum communications: myths and realities

Abstract: Spread spectrum communication techniques date back to the early fifties. Since the earliest applications, system improvements have been more evolutionary than revolutionary. Like most improvements in electronic systems, these are due primarily to the availability of ever higher speed integrated circuit components, which translate in this case to wider spread spectra. In three decades the achievable spreading factor has grown by about three orders of magnitude’ to the point that we are now limited more by bandwidth allocations than by technology limitations. Before we xamine the quantitative effects of spreading, let us catalog briefly the multiple purposes of spread spectrum communications. First, we note that spreading here refers to expansion of the bandwidth well beyond what is required to transmit digital data. Thus, a system transmitting data at a rate ( R ) of 100 Mbits/s using approximately 100 MHz of bandwidth (W) is not spread at all, while a system transmitting at 100 bits/s spread over a spectrum of about 100 MHz has a factor W/R = 106, or 60 dB of so-calledprocessing gain.

A spectral subtraction algorithm for suppression of acoustic noise in speech

Abstract: Spectral subtraction has been shown to be an effective approach for reducing ambient acoustic noise in order to improve the intelligibility and quality of digitally compressed speech. This paper presents a set of implementation specifications to improve algorithm performance and minimize algorithm computation and memory requirements. It is shown spectral subtraction can be implemented in terms of a nonstationary, multiplicative, frequency domain filter which changes with the time varying spectral characteristics of the speech. Using this filter a speech activity detector is defined and used to allow the algorithm to adapt automatically to changing ambient noise environments. Also the bandwidth information of this filter is used to further reduce the residual narrowband noise components which remain after spectral subtraction.

Time-domain algorithms for harmonic bandwidth reduction and time scaling of speech signals

Abstract: Frequency scaling of speech signals by methods based on short-time Fourier analysis (STFA), analytic rooting, and harmonic compression using a bank of filters, is a complex operation which requires a large amount of computation in a digital implementation. It is shown in this paper that, by incorporating pitch frequency information into a frequency-scaling process based on STFA, it is possible, to a good approximation, to perform this operation in the time domain with very few arithmetic operations (one multiplication and two additions per output sample, in most applications). The derivation of the time-domain harmonic scaling (TDHS) algorithms, selection of parameters, and, in particular, the determination of an appropriate weighting function used in the algorithms, as well as several potential applications, are detailed in the paper. Two proposed applications are discussed in greater detail. These are 1) a vocoder system which incorporates waveform coding of the frequency divided signal (by a factor of up to 3), and 2) a computer-based isolated-word recognition system in which all input utterances are compressed to the same duration at the preprocessing phase effecting an overall computation reduction by a factor of up to 3. Computer simulation results which demonstrate the TDHS algorithms' performance are included.

Digital interpolation beamforming for low-pass and bandpass signals

Abstract: Digital time-domain beamforming requires that samples of the sensor signals be available at a sufficient rate to realize accurate time delays for beam steering. For many applications, this input rate, which may be significantly higher than the Nyquist rate required for waveform reconstruction, places stringent requirements on A/D converter hardware and transmission cable bandwidth. Recently, a technique referred to as digital interpolation beamfonning was introduced which greatly relaxes the sampling requirement and provides substantial hardware savings through more flexible design options. In this approach, the sensor channels need only be sampled at a rate which satisfies aliasing requirements. The vernier beam-delay increments are then synthesized using digital interpolation which can be implemented at the beamformer input or output to minimize digital processing complexity. Previously, this concept was presented for the case of "low-pass" signals. This paper extends this work by examining the relationship between interpolation and beamforming for the important class of "bandpass" signals. Specifically, sampling methods are discussed whereby the original waveform can be reconstructed from samples taken at a rate consistent with the bandwidth of the bandpass signal. Beamformer implementations are presented which utilize these bandwidth-sampling techniques in conjunction with interpolation and which compute beam output points at the generally low rate dictated by the signal bandwidth. The interpolation beamformer achieves time-delay quantization (beam-steering accuracy) independent of both the input and output sampling rates. This approach generally requires less hardware than conventional procedures. Interpolation falter characteristics dictated by the bandwidth-sampling procedure are described and efficient methods of implementation employing nonrecursive digital bandpass and low-pass filters are presented.

Design of General Manifold Multiplexers

Abstract: The direct analytical design process for arbitrary multiplexers given in a previous paper is extended to the case of bandpass channel filters connected to a uniform-impedance manifold (e.g., a length of waveguide or transmission line). The previous approximations are greatly improved by adding immittance compensation in a way which not only preserves the canonic form of the network but also assists in the physical construction by spacing the filters along a manifold. The phase shifters between channels are themselves sufficient to compensate the filter interactions to such an extent that contiguous channeling cases are designable. The results are presented mainly in closed form requiring minimal computer optimization. Analysis of multiplexers with frequency-dependent manifolds indicate that there are restrictions on the total bandwidth, but a ten-channel multiplexer is probably feasible, suitable for input and output multiplexers required in typical communications systems. Practical results on a simple manifold triplexer are presented.

Sampling theorems in polar coordinates

Abstract: We investigate the problem of representing an arbitrary class of real functions f(·) in terms of their sampled values along the radius r and at equal angular increments of the azimuthal angle θ. Two different bandwidth constraints on f(r,θ) are considered: Fourier and Hankel. The end result is two theorems which enable images to be reconstructed from their samples. The theorems have potential application in image storage, image encoding, and computer-aided tomography.

Control system design and analysis using closed-loop Nyquist and Bode arrays

Abstract: In this paper a method is described for designing linear multivariable control schemes which have a closed-loop frequency response as close as possible, in a least squares sense, to a desired response. After using characteristic gain loci to ensure system stability, the closed-loop Bode array gives easily understood information about the controlled system in terms of bandwidth, speed of response, resonance and interaction. The closed-loop Nyquist array indicates the robustness of the control scheme for sensor failures ; it also indicates the extent to which state and input noise will be suppressed, since the feedback just multiplies the open-loop disturbances by a unit matrix minus the closed-loop frequency response. Bands of Gershgorin and Ostrowski circles are used to indicate the behaviour for changes in the characteristics of more than one sensor at a time. A similar frequency-response array, obtained by breaking the feedback loops next to the actuators instead of next to the sensors, can be used to p...

Adaptive Array Bandwidth with Tapped Delay-Line Processing

Abstract: The bandwidth of adaptive arrays with tapped delay lines behind the elements is examined. Such processing offers improved bandw over that attainable with quadrature hybrid processing. The performance of a two-element array with four types of processing (equarature hybrids, single delay lines, 3-tap delay lines, and 5-tap delay lines) is compared. It is shown that with half-wavelength element spacing, a quadrature hybrid and single delay-line processor are inadequate at 10-percent bandwidth. A 3-tap processor is adeq however, up to 40-percent bandwidth.

An analog scrambling scheme which does not expand bandwidth, Part I: Discrete time

Abstract: The problem of scrambling a discrete-time analog sequence for the purpose of privacy encoding is studied. A large family of linear orthogonal invertible scrambling transformations is described that result in a negligible expansion of bandwidth and can therefore serve as building blocks in a secure communication system.

An analog scrambling scheme which does not expand bandwidth, Part II: Continuous time

Abstract: The techniques developed in Part I[1] for discrete-time analog scrambling are applied to the problem of scrambling band-limited continuous-time signals or waveforms. The idea behind the waveform scrambler is to sample the waveform (which is assumed to be band-limited) at a rate exceeding the Nyquist rate. The resulting sequence of samples is band-limited in the sense of Part I. The discrete-time scrambler described in Part I is applied to this sequence to produce a nearly band-limited scrambled sequence. A scrambled waveform is formed by modulating the amplitudes of a chain of pulses. This scrambled waveform can be transmitted over a band-limited channel, and the original unscrambled waveform can be recovered at the receiver.

On Quadrature Sampling of Bandpass Signals

Abstract: Deterministic bandpass signals are considered in which the nonzero portions of the signal spectrum are confined to the frequency region 0? ?0 ? ?/2 ? |?| ??0 + ?/2, where ? > 0 is the "bandwidth" of the signal. Quadrature sampling, as introduced by O. D. Grace and S. P. Pitt, requires uniform sampling of both the bandpass signal and its quarter wavelength (based on nominal frequency ?0) translation, each at a common sampling rate depending on the exact relationship between ?0 and ?. When th intersample sample spacing is properly chosen, the bandpass signal can be reconstructed in its entirety from knowledge of the sample values; moreover, with quadrature sampling, the (low-pass) in-phase and quadrature components of the bandpass signal have a simple explicit representation in terms of samples of the original bandpass signal. Time domain techniques, in particular the theory of orthogonal expansions, are here used to derive the quadrature sampling theorem as well as the uniform sampling theorem for bandpass signals, a result usually derived from frequency (spectral) considerations. The resulting minimum sampling rate for the quadrature sampling theorem provides a reduction in the sampling rate previously announced by Grace and Pitt.

Ultrasonic ranging system for a camera

Abstract: An ultrasonic ranging system for a camera includes an ultrasonic transducer responsive to a keying pulse for transmitting a frequency modulated burst of ultrasonic energy toward a subject. The leading half of the burst is a chirp whose frequency varies from a higher to a lower value; during the trailing half of the burst, the frequency remains substantially constant. The system also includes a receiver for processing an echo signal produced by the transducer on receipt of an echo within a predetermined time interval following the burst, and producing a range signal with a characteristic linearly related to the distance of a subject being photographed from the camera. A variable Q filter in the receiver filters echo signals produced by the transducer eliminates the need for a matched filter. The filter Q has a relatively low value during the initial portion of said predetermined time interval causing the bandwidth of the filter to be wide enough to pass all the frequencies of the chirp returned from a subject relatively close to the camera. During the remaining portion of the interval, the Q increases narrowing the filter bandwidth and optimizing it, with regard to the signal-to-noise ratio, for the constant frequency of the burst returned from a relatively remote subject. This arrangement reduces sensitivity of the receiver to echo cancellation due to interference between waves reflected from a nearby subject without adversely affecting the signal-to-noise ratio for returns from remote subjects. An accompanying change in filter admittance reduces angular sensitivity of the receiver to off-axis echoes.

Broadband quasi-taper helical antennas

Abstract: A unique approach is described for widening the bandwidth of a helical antenna with improved gain, pattern, and axial ratio characteristics. The antenna may be described as a nonuniform or quasi-taper helix, which consists of a combination of uniform and tapered helix sections. Measured patterns, gain, axial ratio, and VSWR for various helical antenna configurations are presented and compared. It is shown that a nonuniform quasi-taper helix can provide an operating bandwidth twice that of a conventional uniform-diameter helix.

Time integrating optical signal processing

Abstract: Time integrating acousto-optic processors realize flexible, multipurpose complex signal processing architectures based on correlation algorithms. One- and two-dimensional techniques are presented including examples of spectral analysis and ambiguity function processing. Noncoherent optical processor implementation using interferometric detection with electronic reference is described and experimental results are given.

Spectrum efficiency of a frequency-hopped-DPSK spread-spectrum mobile radio system

Abstract: The spectrum efficiency (bits/s per unit bandwidth) of a frequency-hopped-DPSK spread-spectrum modulation technique proposed for use in urban mobile radio systems is estimated. The analysis is based on simple models for the factors influencing system performance, and requires no computer simulation. It permits straightforward evaluation of spectrum efficiency over a broad range of system parameters. For an isolated service area (cell) with no propagation impairments, the efficiency is 0.12 at a bit error rate of 10-3. When interference from nearby cells is considered (assuming centrally located base stations with omni-directional antennas), the efficiency falls to 0.03 - 0.06, depending on how much performance degradation can be tolerated near the cell boundaries. Finally, the effects of multipath (Rayleigh) fading reduce system efficiency to 0.02 - 0.05.

An electro‐optic tunable filter

Abstract: This letter describes a tunable optical filter that couples light energy between ordinary and extraordinary polarizations in a birefringent electro‐optic crystal at the phase‐matched wavelength by means of a spatially periodic dc electric field. The specific embodiment described here is a LiTaO3 platelet operating with transverse electric field and located between cross polarizers. Analogies to both the Solc birefringent filter and the Harris acousto‐optic filter are cited. The electro‐optic filter is tuned by varying the spatial period of the electric field via an array of separately addressable finger electrodes. We have demonstrated tunability from 4700 A to 4.5 μm. Experimental results of transmittance versus applied voltage and of bandwidth versus pass wavelength agree well with theory. The flexibility of this type of electronic filter in allowing synthesis of arbitrary transfer functions through nonperiodic voltage distributions is emphasized.

Variable bandwidth filtering and frequency converting system

Abstract: A system which provides frequency conversion and continuously variable bandwidth control uses fixed filters to which opposite edges of the passband of the signal are applied in inverted frequency relationship. A channel contains the filters and three mixers to which injection signals derived from a single variable frequency oscillator and a fixed oscillator are applied to vary the bandwidth of the system without affecting the center frequency of the signal after processing in the system. The system may be designed to obtain either a variable bandpass or rejection band filtering characteristic.

Segmented chirp waveform implemented radar system

Abstract: In a radar system utilizing same, a segmented chirp waveform is built up by frequency hopping trains of simple chirp pulses. Each chirp pulse in a train is identical and is transmitted on one of a set of discrete carrier frequencies. The carrier frequencies are uniformly spaced in a predetermined bandwidth and are used in linear sequence. After transmitting on the last carrier frequency, the entire pattern is repeated, starting again at the first carrier frequency. The returns from the individual pulses are digitized and stored and the history thereof is then assembled by a digital signal processor into a high resolution image.

Miniature surface-acoustic-wave filters

Abstract: An introduction to Surface-Acoustic-Wave (SAW) devices-their operation, advantages, and applications--is presented. The process of design and realization of high-performance SAW filters is examined from initial specifications to final device operation. Several techniques drawn from a variety of disciplines illustrate the implementation of finite impulse response SAW frequency functions. Material properties, transducer weighting, and computer-aided design are treated. An example of the design and testing of a 321-MHz 0.35-percent bandwidth, 65-dB sidelobe, 15-dB insertion loss filter is presented.

Noise reduction system

Abstract: A process and apparatus (a) for reducing the perceived noise without perceptible loss of fidelity upon aural reproduction, of an audio input signal in electrical form by means of an electronic circuit wherein the upper limit of the bandwidth of the signal is automatically and continuously adjusted in accordance with the relationship of (i) the peak amplitude of the rate of change of the signal with respect to time to (ii) the instantaneous amplitude within a selected frequency band of the signal, and (b) a system for detecting and discriminating between noise and audio transients in the input signal and retaining the audio transients in the output signal.

Acoustooptic spectrum analysis: a high performance hybrid technique

Abstract: A hybrid technique for wide-bandwidth high-resolution acoustooptic spectrum analysis is described that combines features of 2-D space-integrating and time-integrating techniques. Performance features include extended small-signal detectability, improved optical efficiency, and insensitivity to high frequency laser noise. An optically transformed periodic chirp provides a distributed local oscillator that permits wide system bandwidth.

Pseudorandom Code Waveform Design for CW Radar

Abstract: Certain surveillance radar applications require that the transmitted signal be continuous wave (CW) and wide bandwidth to meet range and range-rate resolution, and also performance requirements in clutter, multiple targets, or other interfering environments. Although an unmodulated CW radar is well suited for making range-rate (Doppler) measurements, a modulation of the transmitted signal is generally required to make range measurements. We describe the use of a pseudorandom code (PRC) which modulates the radar transmitter carrier to meet the range resolution and measurement requirements. The tradeoffs for selecting the PRC code parameters are presented together with a functional description. It is shown that in order to achieve the desired performance, it is necessary that Doppler ambiguities be avoided. This generally causes the range measurement to be ambiguous. The range ambiguity can be removed by staggering the code clock frequency during the time on target. Special emphasis is placed on an algorithm which allows range measurements to be made to an accuracy of one code bit.

Bandwidth Efficiency and Probability of Error Performance of MSK and Offset QPSK Systems

Abstract: In this paper a model of Minimum Shift Keying (MSK) is introduced, an analysis of which shows that the theoretical maximum bandwidth efficiency of MSK is 2 bits/s/Hz, the same as for Quadrature Phase Shift Keying (QPSK) and Offset Quadrature Phase Shift Keying (Offset QPSK). The filtering necessary to achieve minimum spectrum width as well as a raised cosine spectrum while maintaining the condition of zero intersymbol interference for Offset QPSK is reviewed, and the equivalent filtering for MSK derived. In addition, partioning of this filtering between transmitter and receiver for optimum probability of error (P e ) versus receiver input signal to noise ( S/N ) ratio is given. Computer simulation results are presented for a MSK system and an Offset QPSK system, both filtered for 0%, 50% and 100% excess bandwidth raised cosine spectrum. The results show fully open eyes at the decision threshold input for all cases. They also show that when filtered to give the same spectrum the different system have identical P e versus S/N performance, S/N being defined at the input to the receiver decision threshold, i.e., after all the filtering for the desired raised cosine spectrum. This performance is best for 100% excess bandwidth filtering and worst for 0% excess bandwidth filtering.

Multichannel frequency translation of sampled waveforms by decimation and interpolation

Abstract: A system and method for frequency division demultiplexing a received broad band signal into km channels, each capable of containing a baseband signal component and comprising m descending tiers of identical channel division modules with the first tier thereof comprising a first module responsive to the received broad band signal to divide such broad band signal (at baseband) into k new channels each containing a baseband spectral component. Each module of each tier of modules is constructed to respond individually (by sampling and de-sampling techniques) to the spectral component in individual ones of the channels outputted from the immediately preceding higher order tier of modules to form k new additional channels each containing a baseband spectral component. Each baseband spectral component has an upper bandwidth limit which bears the same ratio to its sampling rate as the upper bandwidth limit of the originally received broad band signal bears to its sampling rate.

Digital signal processing method and apparatus

Abstract: Encoded digital television signals having a defined bandwidth are transmitted at reduced bandwidth by regularly omitting (18) one in every n of the signal samples, where n is greater than two and is preferably three or four. At a receiver the omitted samples are regenerated by estimating (26) their values from the samples of the reduced-rate transmitted signal by making use of frequency components outside the normal defined bandwidth of the original signal. The estimation can be achieved by a symmetrical digital transversal filter of which one in every n coefficients is zero, the amplitude/frequency characteristics of the filter being unity throughout the defined bandwidth of the TV signal and antisymmetric about half the mean lower sample rate.

Some techniques for evaluating the bandwidth characteristics of adaptive nulling systems

Abstract: Some techniques for evaluating the bandwidth characteristics of adaptive nulling systems are presented. The system performance is categorized according to its antenna characteristics and the post-antenna channel tracking characteristics. The former tend to be radio-frequency (RF) percentage bandwidth dependent and the latter on the achievable component tolerances used to implement the channel (whether at RF or some lower intermediate frequency (IF)). Measured results are also presented for a seven-beam paraboloid reflector-type multiple-beam antenna (MBA), and the effects of feed-reflector multipath inherent in this geometry on the cancellation bandwidth is evaluated in detail.