Acceptable Level Of Distortion Research Articles

Event Distortion-Based Clustering Algorithm for Energy Harvesting Wireless Sensor Networks

Wireless sensor networks (WSNs) consist of compact deployed sensor nodes which collectively report their sensed readings about an event to the Base Station (BS). In WSNs, due to the dense deployment, sensor readings can be spatially correlated and it is nonessential to transmit all their readings to the BS. Therefore, for more energy efficient, it is vital to choose which sensor node should report their sensed readings to the BS. In this paper, the event distortion-based clustering (EDC) algorithm is proposed for the spatially correlated sensor nodes. Here, the sensor nodes are assumed to harvest energy from ambient electromagnetic radiation source. The EDC algorithm allows the energy-harvesting sensor nodes to select and eliminate nonessential nodes while maintain an acceptable level of distortion at the BS. To measure the reliability, a theoretical framework of the distortion function is first derived for both single-hop and two-hop communication scenarios. Then, based on the derived theoretical framework, the EDC algorithm is introduced. Through extensive simulations, the performance of the EDC algorithm is evaluated in terms of achievable distortion level, number of alive nodes and harvested energy levels. As a result, EDC algorithm can successfully exploit both the spatial correlation and energy harvesting to improve the energy efficiency while preserving an acceptable level of distortion. Furthermore, the performance comparisons reveal that the two-hop communication model outperforms the single-hop model in terms of the distortion and energy-efficiency.

Measuring Secrecy by the Probability of a Successful Guess

The secrecy of a communication system in which both the legitimate receiver and an eavesdropper are allowed some distortion is investigated. The secrecy metric considered is the exponent of the probability that the eavesdropper estimates the source sequence successfully within an acceptable distortion level. The problem is first studied when the transmitter and the legitimate receiver do not share any key and the transmitter is not subject to a rate constraint, which corresponds to a stylized model of a side channel and reveals connections to source coding with side information. The setting is then generalized to include a shared secret key between the transmitter and the legitimate receiver and a rate constraint on the transmitter, which corresponds to the Shannon cipher system. A single-letter characterization of the highest achievable exponent is provided, and asymptotically-optimal strategies for both the primary user and the eavesdropper are demonstrated.

Analysis and Implementation of a Time-Interleaved ADC Array for a Software-Defined UWB Receiver

A software-defined radio (SDR) for ultrawideband (UWB) communication systems places several stringent requirements on the analog-to-digital converter (ADC). One alternative to using a single ADC is to sample the received signal with an array of lower speed ADCs that were driven by interleaved sampling clocks; however, mismatches among the ADCs will result in signal distortion. This paper makes three important contributions to overcoming this problem: 1) analytical quantification of the impact of ADC gain, offset, and timing mismatches on the performance of a time-interleaved sampling ADC array for UWB signals; 2) demonstration of the efficacy of using a pilot-based matched-filter architecture to mitigate the impact of timing mismatches in the presence of multipath; and 3) implementation of an 8-ADC time-interleaved UWB SDR testbed that operates at an effective sampling frequency of 6.4 GHz. In addition, our findings allow for the design specification of the number of pilots required to obtain a desired system performance. The simulation and measured performance results from this paper demonstrate that ADC mismatches can be controlled to within plusmn10%, yielding acceptable levels of distortion and bit-error-rate (BER) performance on the UWB SDR testbed. Both analytical and simulation results also demonstrate the efficacy of a pilot-based matched filter in mitigating the impact of timing mismatch errors, even in the presence of multipath.

Optimal Scheduling of LTC and Shunt Capacitors in Large Distorted Distribution Systems Using Evolutionary-Based Algorithms

The optimal scheduling of load tap changers (LTCs) and switched shunt capacitors for simultaneously minimizing energy loss and improving voltage profile while taking harmonics into account is performed using evolutionary-based algorithms (EAs). The proposed algorithm is capable of optimizing large distribution systems with different types of nonlinear loads. The decouple approach is employed for harmonic power-flow calculations, while two EAs are developed to determine the load interval division and near-optimal schedule. The inclusion of harmonics provides the optimization benefits while maintaining the acceptable distortion levels. The scheduling is carried out for the IEEE 123-bus with 14 shunt capacitors and 12 nonlinear loads. The main contributions are inclusion of harmonics in the optimal scheduling problem and its application to large distribution systems with multiple nonlinear loads.

A Knapsack Model for Bandwidth Management of Prerecorded Multiple MPEG Video Sources

In this article we provide a framework for controlling the bit rate of multiple prerecorded MPEG video sequences by choosing the quantization factors assigned to individual sources in a way that the total mean square error at the output of the encoder is minimized. We propose and test a knapsack model for the selection of the quantization factors. Our computations based on a set of relatively diverse video sequences reveal that the proposed model achieves a high utilization of the available bandwidth and acceptable distortion levels without any data loss.

High-quality low-complexity wavelet-based compression algorithm for audio signals

Wavelets have recently emerged as a powerful tool for signal compression, particularly in the areas of image, video, and audio compression. In this paper, we present a low-complexity wavelet-based audio compression algorithm that is capable of handling fairly arbitrary audio sources. The algorithm transforms the incoming audio data into the wavelet domain, and compresses data by exploring redundancy in the wavelet coefficients and exploiting the large runs of zeros in the transformed signal. Also there is a possibility of applying a threshold to the non-zero coefficients, thus a further increase in the number of zeros is expected. The audio signal is first preprocessed to scale down the wavelet coefficients. Then the preprocessed signal is wavelet transformed using a bi-orthogonal discrete wavelet transform (DWT) and threshold by applying energy compaction strategy. Encoding represents the threshold coefficients in compact form. A new encoding technique that is easy to implement, and that provides a reasonable compression ratio for a certain acceptable distortion level has been developed to encode the threshold DWT. So, a bit rate can be controlled such that the algorithm operates at virtually any pre-selected bit rate. The motivation of using the bi-orthogonal wavelet transform is that it permits the use of a much broader class of filters, and this class includes symmetric linear phase filters. The superior performance of this algorithm is also demonstrated by comparing it with two other popular audio compression techniques and this meets the requirements of multimedia computing.

The effect of storage time on the accuracy and dimensional stability of reversible hydrocolloid impression material

Statement of Problem. It has been stated that reversible hydrocolloid impression material must be poured immediately to remain accurate. This may not be true with modern formulations of reversible hydrocolloid. The accuracy of the impression, if kept at 100% humidity for varying periods, may not be affected as critically as has been suspected. Purpose. The purpose of this study was to evaluate the effect of storage time in a 100% humidity environment on the accuracy of gypsum casts poured from a newer formulation of reversible hydrocolloid impressions. Material and Methods. With the use of a stainless steel master model, 5 standardized reversible hydrocolloid impressions were made for each time interval tested (n = 5). Impressions were made with the use of both Cartiloid blue syringe and Slate tray reversible hydrocolloid material (both manufactured by Van R) to simulate a common clinical protocol. Six groups, representing times at 0, 30, 60, 120, 180, and 210 minutes, were evaluated for a total of 30 specimens. Gypsum casts were measured with a traveling microscope, and dimensional changes were compared between groups with ANOVA and Tukey intervals (α=.05). Results. Measurements made up to and including the 60-minute storage time showed clinically acceptable distortion levels. These measurements were similar to the metal model and the gypsum casts poured at time 0 minutes. The only statistically significant difference between the gypsum cast and the metal master model was the distance between abutments at storage time 180 minutes (master model = 30.04 mm [SD 0.001]; gypsum casts = 29.944 mm [SD 0.076]). This difference was −91 μm (P=.002). In general, the gypsum abutments became wider and shorter with time. There was no significant difference between any gypsum abutment in regard to width or height when compared with the metal master model for all the storage times tested. However, as time increased, measurement became harder to accomplish, leading to a larger standard deviation and less reliable results. The distance between abutments remained relatively constant for storage times up to and including 60 minutes. For times longer than 60 minutes, the distance between abutments was reduced and more difficult to read. The distance between abutments at the 3-hour storage time was significantly different than the metal standard and time 0 gypsum casts. Conclusion. If a discrepancy of 78 mm (twice the 39 mm single margin opening) is clinically acceptable, then the results of this study show that this type of reversible hydrocolloid can be stored in 100% humidity for at least 60 minutes. (J Prosthet Dent 2001;86:244-50.)

A two codebook format for robust quantization of line spectral frequencies

An important problem in speech coding is the quantization of linear predictive coefficients (LPCs) with the smallest possible number of bits while maintaining robustness to a large variety of speech material and transmission media. Since direct quantization of LPCs is known to be unsatisfactory, the authors consider this problem for an equivalent representation, namely, the line spectral frequencies (LSFs). To achieve an acceptable level of distortion a scalar quantizer for LSFs requires a 36 bit codebook. The authors derive a 30 bit two-quantizer scheme which achieves a performance equivalent to this scalar quantizer. The two-quantizer format consists of both a vector and a scalar quantizer such that for each input, the better quantizer is used. The vector quantizer is designed from a training set that reflects the joint density (for coding efficiency) and which ensures coverage (for robustness). The scalar quantizer plays a pivotal role in dealing better with regions of the space that are sparsely covered by its vector quantizer counterpart. A further reduction of 1 bit is obtained by formulating a new adaptation algorithm for the vector quantizer and doing a dynamic programming search for both quantizers. The method of adaptation takes advantage of the ordering of the LSFs and imposes no overhead in memory requirements. The dynamic programming search is feasible due to the ordering property. Subjective tests in a speech coder reveal that the 29 bit scheme produces equivalent perceptual quality to that when the parameters are unquantized. >

Selecting the gain for radio microphone (FM) systems: theoretical considerations and practical limitations.

This paper provides theoretical estimates of the signal-to-noise (S/N) ratio advantage that can be gained from a radio microphone (FM) system used with an environmental (EV) microphone, compared with the S/N ratio delivered by a conventional hearing aid alone. These estimates show that the S/N advantage gained is a function of the speaker/listener distance and the FM/EV gain difference, the greater the distance and the closer the FM gain is to the EV gain, the greater the S/N advantage. The implications for choice of FM and EV gain when fitting FM systems are discussed. The two goals of equating the FM and EV outputs and maximizing the S/N ratio are mutually incompatible. It is concluded that the FM gain should be set to be 10dB less than the EV gain unless the situation can be closely monitored, in which case the FM gain should be set as high as possible consistent with loudness tolerance and acceptable levels of distortion.

A Microprocessor-Based Controller for Static VAr-Compensator

The optimum ratings of static reactive power compensators connected to incustrial distribution networks providing simultaneously several tasks can be determined only having complex information about the supply network to be compensated. the consumers fed from it and the acceptable distortion level. These factors influencing the size of the compensator are of varying nature, e.g. because of the changing of the feeding point. By means of a suitably “intelligent” control unit, an oversizing of the reactive power compensator can be avoided and its installed power can be utilized more efficiently. the paper describes a microprocessor-based control unit that permits by its flexibility a relatively simple and reliable achievement of the above task. The Intel 8031 microprocessor performs the calculation of the reactive powers and the control algorithm as well as the firing of the thyristors.

On "Distortion of Single-Sideband Transmission Due to AM/PM Conversion"

In a previous paper, this author developed an expression for the intermodulation distortion generated in single-sideband (SSB), amplitude modulation (AM) systems carrying frequency-division multiplex (FDM) telephony, due to AM/PM conversion. For the purpose of the theory, the SSB signal is replaced by an equivalent system consisting of a carrier simultaneously phase and amplitude modulated. To simplify the analysis, the AM/PM conversion factor was assumed constant over the significant range of variation of the envelope of the effective carrier. This is not, in fact, a very realistic assumption since the AM/PM factor varies quite, considerably with the input power level. Particularly in the case of a traveling wave tube working well below saturation, the dependence is likely to be linear It has subsequently been realized that the analysis is still quite tractable when the AM/PM factor is proportional to the input power level. In fact, on this assumption, it turns out that the expression given previously for the intermodulation distortion (expression (14) of Medhurst et al. [ibid., vol. CS-12, pp. 166-176, June 1964]) applies, if the AM/PM conversion factor is given its value at the rms level of the multichannel SSB signal. This is a useful result, since it appears that for acceptable distortion levels in TWT's used in satellite system ground stations, for example, it would be necessary to operate the tube in this linear region. The analysis is presented.