TL;DR: To improve radio sensitivity of the sensing function through processing gain, three digital signal processing techniques are investigated: matched filtering, energy detection and cyclostationary feature detection.
Abstract: There are new system implementation challenges involved in the design of cognitive radios, which have both the ability to sense the spectral environment and the flexibility to adapt transmission parameters to maximize system capacity while coexisting with legacy wireless networks. The critical design problem is the need to process multigigahertz wide bandwidth and reliably detect presence of primary users. This places severe requirements on sensitivity, linearity and dynamic range of the circuitry in the RF front-end. To improve radio sensitivity of the sensing function through processing gain we investigated three digital signal processing techniques: matched filtering, energy detection and cyclostationary feature detection. Our analysis shows that cyclostationary feature detection has advantages due to its ability to differentiate modulated signals, interference and noise in low signal to noise ratios. In addition, to further improve the sensing reliability, the advantage of a MAC protocol that exploits cooperation among many cognitive users is investigated.
TL;DR: A memory polynomial model for the predistorter is proposed and implemented using an indirect learning architecture and linearization performance is demonstrated on a three-carrier WCDMA signal.
Abstract: Power amplifiers (PAs) are inherently nonlinear devices and are used in virtually all communications systems. Digital baseband predistortion is a highly cost-effective way to linearize PAs, but most existing architectures assume that the PA has a memoryless nonlinearity. For wider bandwidth applications such as wideband code-division multiple access (WCDMA) or wideband orthogonal frequency-division multiplexing (W-OFDM), PA memory effects can no longer be ignored, and memoryless predistortion has limited effectiveness. In this paper, instead of focusing on a particular PA model and building a corresponding predistorter, we focus directly on the predistorter structure. In particular, we propose a memory polynomial model for the predistorter and implement it using an indirect learning architecture. Linearization performance is demonstrated on a three-carrier WCDMA signal.
TL;DR: A quantitative comparison of both approaches to spectrum pooling aims at enabling public access to these spectral ranges without sacrificing the transmission quality of the actual license owners, and it is obvious that both approaches sacrifice bandwidth of the rental system.
Abstract: The public mobile radio spectrum has become a scarce resource while wide spectral ranges are only rarely used. Here, the new strategy called spectrum pooling is considered. It aims at enabling public access to these spectral ranges without sacrificing the transmission quality of the actual license owners. Unfortunately, using OFDM modulation in a spectrum pooling system has some drawbacks. There is an interaction between the licensed system and the OFDM based rental system due to the non-orthogonality of their respective transmit signals. This interaction is described mathematically, providing a quantitative evaluation of the mutual interference that leads to an SNR loss in both systems. However, this interference can be mitigated by windowing the OFDM signal in the time domain or by the adaptive deactivation of adjacent subcarriers providing flexible guard bands between licensed and rental system. It is obvious that both approaches sacrifice bandwidth of the rental system. A quantitative comparison of both approaches is given as a tradeoff between interference reduction and throughput in the rental system.
TL;DR: In this paper, the author illustrates why telecommunications data rates are as predictable as Moore's Law and illustrates why it is possible to predict the rate of data acquisition in the USA asymptotically.
Abstract: The author illustrates why telecommunications data rates are as predictable as Moore's Law.
TL;DR: A repetitive-based controller for active power filters, which compensates selected current harmonics produced by distorting loads using a closed-loop repetitive- based control scheme based on a finite-impulse response digital filter, which allows full compensation of selected frequencies, even if the active filter has limited bandwidth.
Abstract: This paper proposes a repetitive-based controller for active power filters, which compensates selected current harmonics produced by distorting loads. The approach is based on the measurement of line currents and performs the compensation of selected harmonics using a closed-loop repetitive-based control scheme based on a finite-impulse response digital filter. Compared to conventional solutions based on stationary-frame current control, this approach allows full compensation of selected frequencies, even if the active filter has limited bandwidth. Compared to synchronous-frame harmonic regulations on line currents, the complexity of the proposed algorithm is independent of the number of compensated harmonics. Moreover, it is more appropriate for digital signal processor implementation and less sensitive to rounding and quantization errors when finite word length or fixed-point implementation is considered. Experimental results on a 5-kVA prototype confirm the theoretical expectations.
TL;DR: In this article, a configurable vibration sensor consisting of a sensor circuit, an analog-to-digital converter and a processor is presented, where each sensor circuit comprises a vibration sensing element and a variable bandwidth filter controllable by the processor.
Abstract: A configurable vibration sensor comprising a sensor circuit, an analog-to-digital converter and a processor, where each sensor circuit comprises a vibration sensing element and a variable bandwidth filter controllable by the processor. In addition to the variable bandwidth filter, other configurable elements may also be employed in the sensor circuit, including a variable gain amplifier. These configurable elements allow the configurable vibration sensor to be configured for different vibration measurement applications when measuring vibrations from vibrating structures such as machinery and the like.
TL;DR: Some plug-in type of bandwidth selectors, which are based on non-parametric estimation of an approximation of the mean integrated squared error, are proposed and it is concluded that an appropriately chosen PI bandwidth selector and the BT bandwidth selector perform comparably and both outperform the CV bandwidth.
TL;DR: In this paper, a broadband power combining device consisting of a wedge-shaped metal carrier, an input antipodal finline structure, one or more active elements, and a corresponding output antenna array is presented.
Abstract: A broadband power combining device includes an input port, an input waveguide section, a center waveguide section formed by stacked wedge-shaped trays, an output waveguide section, and an output port. Each tray is formed of a wedge-shaped metal carrier, an input antipodal finline structure, one or more active elements, an output antipodal finline structure, and attendant biasing circuitry. The wedge-shaped metal carriers have a predetermined wedge angle and predetermined cavities. The inside and outside surfaces of the metal carriers and surfaces of the cavity all have cylindrical curvatures. When the trays are assembled together, a cylinder is formed defining a coaxial waveguide opening inside. The antipodal finline structures form input and output arrays. An incident EM wave is passed through the input port and the input waveguide section, distributed by the input antipodal finline array to the active elements, combined again by the output antipodal finlines array, then passed to the output waveguide section and output port. A hermetic sealing scheme, a scheme for improving the power combining efficiency and thermal management scheme are also disclosed. The broadband power combining device operates with multi-octave bandwidth and is easy to manufacture, well-managed thermally, and highly efficient in power combining.
TL;DR: A wide-bandwidth continuous-time sigma-delta ADC is implemented in a 0.13-/spl mu/m CMOS circuit that achieves a dynamic range of 11 bits over a bandwidth of 15 MHz.
Abstract: A wide-bandwidth continuous-time sigma-delta ADC is implemented in a 0.13-/spl mu/m CMOS. The circuit is targeted for wide-bandwidth applications such as video or wireless base-stations. The active blocks are composed of regular threshold voltage devices only. The fourth-order architecture uses an OpAmp-RC-based loop filter and a 4-bit internal quantizer operated at 300-MHz clock frequency. The converter achieves a dynamic range of 11 bits over a bandwidth of 15 MHz. The power dissipation is 70 mW from a 1.5-V supply.
TL;DR: In this paper, a /spl Sigma/spl Delta/ fractional-N frequency synthesizer targeting WCDMA receiver specifications is presented, where spurs compensation and linearization techniques, the PLL bandwidth is significantly extended with only a slight increase in the integrated phase noise.
Abstract: A /spl Sigma//spl Delta/ fractional-N frequency synthesizer targeting WCDMA receiver specifications is presented. Through spurs compensation and linearization techniques, the PLL bandwidth is significantly extended with only a slight increase in the integrated phase noise. In a 0.18-/spl mu/m standard digital CMOS technology a fully integrated prototype with 2.1-GHz output frequency and 35 Hz resolution has an area of 3.4 mm/sup 2/ PADs included, and it consumes 28 mW. With a 3-dB closed-loop bandwidth of 700 kHz, the settling time is only 7 /spl mu/s. The integrated phase noise plus spurs is -45 dBc for the first WCDMA channel (1 kHz to 1.94 MHz) and -65 dBc for the second channel (2.5 to 6.34 MHz) with a worst case in-band (unfiltered) fractional spur of -60 dBc. Given the extremely large bandwidth, the synthesizer could be used also for TX direct modulation over a broad band. The choice of such a large bandwidth, however, still limits the spur performance. A slightly smaller bandwidth would fulfill WCDMA requirements. This has been shown in a second prototype, using the same architecture but employing an external loop filter and VCO for greater flexibility and ease of testing.
TL;DR: A fast, easy-toimplement algorithm for localizing a source using received signal strength measurements is presented, based on incremental subgradient optimization methods, that is much lower than that used by other schemes, especially as network size grows.
Abstract: This paper describes a new approach to the source localization and tracking problem in wireless sensor networks. A fast, easy-toimplement algorithm for localizing a source using received signal strength measurements is presented. The algorithm is based on incremental subgradient optimization methods. Using theory on the convergence rates of these methods we characterize the amount of in-network communication required to achieve an accurate estimate of the source’s location. In comparison to other localization and tracking algorithms described in the literature, the amount of communication (and thus energy and bandwidth) used by our algorithm is much lower than that used by other schemes, especially as network size grows.
TL;DR: In this article, a high-speed CMOS adaptive cable equalizer using an enhanced low-frequency gain control method is described, which alleviates the speed bottleneck of the conventional adaptation method.
Abstract: This paper describes a high-speed CMOS adaptive cable equalizer using an enhanced low-frequency gain control method. The additional low-frequency gain control loop enables the use of an open-loop equalizing filter, which alleviates the speed bottleneck of the conventional adaptation method. In addition, combined adaptation of low-frequency gain and high-frequency boosting improves the adaptation accuracy while supporting high-frequency operation. The open-loop equalizing filter incorporates a merged-path topology and offers infinite input impedance, which are suitable for higher frequency operation and cascaded design. This equalizing filter controls its common-mode output voltage level in a feedforward manner, thereby improving bandwidth. A prototype chip was fabricated in 0.18-/spl mu/m four-metal mixed-mode CMOS technology. The realized active area is 0.48/spl times/0.73 mm/sup 2/. The prototype adaptive equalizer operates up to 3.5 Gb/s over a 15-m RG-58 coaxial cable with 1.8-V supply and dissipates 80 mW. Moreover, the equalizing filter in manual adjustment mode operates up to 5 Gb/s over a 15-m RG-58 coaxial cable.
TL;DR: Small-signal frequency- and Laplace-domain models for the different types of uniformly-sampled pulse-width modulators are derived theoretically and the results obtained are verified by means of experimental data retrieved from a test setup.
Abstract: As the performance of digital signal processors has increased rapidly during the last decade, there is a growing interest to replace the analog controllers in low power switching converters by more complicated and flexible digital control algorithms. Compared to high power converters, the control loop bandwidths for converters in the lower power range are generally much higher. Because of this, the dynamic properties of the uniformly-sampled pulse-width modulators used in low power applications become an important restriction for the maximum achievable bandwidth of control loops. After the discussion of the most commonly used uniformly-sampled pulse-width modulators, small-signal frequency- and Laplace-domain models for the different types of uniformly-sampled pulse-width modulators are derived theoretically. The results obtained are verified by means of experimental data retrieved from a test setup.
TL;DR: Analysis of the Cramer-Rao bound (CRB) on source localization accuracy reveals that a distributed processing scheme involving bearing estimation at the individual arrays and time-delay estimation between sensors on different arrays performs nearly as well as the optimum scheme while requiring less communication bandwidth with a central processing node.
Abstract: Multiple sensor arrays provide the means for highly accurate localization of the (x,y) position of a source. In some applications, such as microphone arrays receiving aeroacoustic signals from ground vehicles, random fluctuations in the air lead to frequency-selective coherence losses in the signals that arrive at widely separated sensors. We present performance analysis for localization of a wideband source using multiple, distributed sensor arrays. The wavefronts are modeled with perfect spatial coherence over individual arrays and frequency-selective coherence between distinct arrays, and the sensor signals are modeled as wideband, Gaussian random processes. Analysis of the Cramer-Rao bound (CRB) on source localization accuracy reveals that a distributed processing scheme involving bearing estimation at the individual arrays and time-delay estimation (TDE) between sensors on different arrays performs nearly as well as the optimum scheme while requiring less communication bandwidth with a central processing node. We develop Ziv-Zakai bounds for TDE with partially coherent signals in order to study the achievability of the CRB. This analysis shows that a threshold value of coherence is required in order to achieve accurate time-delay estimates, and the threshold coherence value depends on the source signal bandwidth, the additive noise level, and the observation time. Results are included based on processing measured aeroacoustic data from ground vehicles to illustrate the frequency-dependent signal coherence and the TDE performance.
TL;DR: In this paper, the authors summarized the topics of array processing for wideband signals in smart antenna-based applications, and proposed two main approaches based on the CSS method and beam-space processing has been introduced.
Abstract: In this article, we summarized the topics of array processing for wideband signals in smart antenna-based applications. For wideband beamforming, the TDF1B and FDFIB methods can provide the frequency-invariant beam-pattern over an arbitrarily wide bandwidth. The FDFIB method is applicable with arbitrary antenna arrays and suitable for switched beams and direction-finding approaches. The frequency-invariant beam-pattern can be designed with prescribed narrow main beam width and low SLL over a wide bandwidth by using two FDFIBs in a spatial interpolation process. For wideband DOA estimation, two main approaches based on the CSS method and beam-space processing has been introduced. The latest approach can provide lower resolution threshold, lower RMSE of estimate, and lower computational complexity. Finally, practical considerations in implementation of an antenna array regarding to array errors and mutual coupling have been considered. In the presence of array errors and mutual coupling, the performance of wideband DOA estimation is strongly degraded.
TL;DR: A novel parametric approach for constructing families of intersymbol-interference (ISI)-free pulses is presented and examined, and a number of theorems that relate time-domain behaviors of a pulse to the pulse's frequency spectrum are proved.
Abstract: A novel parametric approach for constructing families of intersymbol-interference (ISI)-free pulses is presented and examined. Some new pulses so constructed have smaller maximum distortion, a more open receiver eye, and a smaller probability of error in the presence of symbol-timing error than the Nyquist raised-cosine pulse for the same excess bandwidth. The parametric approach gives more degrees of freedom in the design of ISI-free pulses, and subsumes previous ISI-free pulses as special cases. A number of theorems that relate time-domain behaviors of a pulse to the pulse's frequency spectrum are proved. A previously known result relating pulse tail-time decay to discontinuity of the pulse-frequency spectrum is corrected and clarified.
TL;DR: The paper addresses a LQG optimal control problem involving bit-rate communication capacity constraints with a discrete-time partially observed system perturbed by white noises and shows that where the estimator-coder separation principle holds, the controller- coder one fails to be true.
TL;DR: In this paper, the authors present a framework for bandwidth extension for low-frequency audio signals, based on physics and psychophysics, with a focus on the effect of pitch and amplitude.
Abstract: Preface.I Introduction.I.1 Bandwidth Defined.I.2 Historic Overview.I.2.1 Electroacoustic Transducers.I.2.2 Sound Quality.I.3 Bandwidth Extension Framework.I.3.1 Introduction.I.3.2 The Framework.1 From Physics to Psychophysics.1.1 Signal Theory.1.1.1 Linear and Non-linear Systems.1.1.2 Continuous-time LTI (LTC) Systems.1.1.3 Discrete-time LTI (LTD) Systems.1.1.4 Other Properties of LTI Systems.1.1.5 Digital Filters.1.2 Statistics of Audio Signals.1.2.1 Speech.1.2.2 Music.1.3 Loudspeakers.1.3.1 Introduction to Acoustics.1.3.2 Loudspeakers.1.3.3 Bessel and Struve Functions.1.4 Auditory Perception.1.4.1 Physical Characteristics of the Peripheral Hearing System.1.4.2 Non-linearity of the Basilar Membrane Response.1.4.3 Frequency Selectivity and Auditory Filters.1.4.4 Loudness and Masking.1.4.5 Pitch.1.4.6 Timbre.1.4.7 Auditory Scene Analysis.1.4.8 Perceptual Modelling - Auditory Image Model.2 Psychoacoustic Bandwidth Extension for Low Frequencies.2.1 Introduction.2.2 Psychoacoustic Effects for Low-frequency Enhancement of Small Loudspeaker Reproduction.2.2.1 Pitch (Harmonic Structure).2.2.2 Timbre (Spectral Envelope).2.2.3 Loudness (Amplitude) and Tone Duration.2.3 Low-Frequency Psychoacoustic Bandwidth Extension Algorithms.2.3.1 Overview.2.3.2 Non-Linear Device.2.3.3 Filtering.2.3.4 Gain of Harmonics Signal.2.4 Low-Frequency Psychoacoustic Bandwidth Extension with Frequency Tracking.2.4.1 Non-Linear Device.2.4.2 Frequency Tracking.2.5 Subjective Performance of Low-Frequency Psychoacoustic Bandwidth Extension Algorithms.2.5.1 'Virtual Bass'.2.5.2 'Ultra Bass'.2.6 Spectral Characteristics of Non-Linear Devices.2.6.1 Output Spectrum of a Rectifier.2.6.2 Output Spectrum of Integrator.2.6.3 Output Spectra in Discrete Time.2.6.4 Output Spectrum of Clipper.3 Low-frequency Physical Bandwidth Extension.3.1 Introduction.3.2 Perceptual Considerations.3.2.1 Pitch (Spectral Fine Structure).3.2.2 Timbre (Spectral Envelope).3.2.3 Loudness (Amplitude).3.3 Low-frequency Physical Bandwidth Extension Algorithms.3.3.1 Systems with Low-frequency Extension.3.3.2 Non-linear Device.3.3.3 Filtering.3.3.4 Gain of Harmonics Signal.3.4 Low-frequency Physical Bandwidth Extension Combined with Low-frequency Psychoacoustic Bandwidth Extension.4 Special Loudspeaker Drivers for Low-frequency Bandwidth Extension.4.1 The Force Factor.4.2 High Force Factor Drivers.4.3 Low Force Factor Drivers.4.3.1 Optimal Force Factor.4.4 Transient Response.4.4.1 Gated Sinusoid Response.4.4.2 Impulse Response.4.5 Details of Lumped-element Parameters and Efficiency.4.6 Discussion.5 High-frequency Bandwidth Extension for Audio.5.1 Introduction.5.2 The Limits of Deconvolution.5.3 Perceptual Considerations.5.3.1 Pitch (Harmonic Structure).5.3.2 Timbre (Spectral Envelope).5.3.3 Loudness (Amplitude).5.3.4 Effects of Hearing Loss.5.3.5 Conclusions.5.4 High-frequency Bandwidth Extension for Audio.5.4.1 Non-linear Device.5.4.2 Filtering.5.4.3 Gain of Harmonics Signal.5.5 Spectral Band Replication (SBR).5.6 High-frequency Bandwidth Extension by Instantaneous Compression.5.6.1 Introduction and Algorithm.5.6.2 Analysis of Harmonics Generation.5.6.3 Implementation.5.6.4 Examples.5.6.5 Approximation of the Function tanh(Z).6 Bandwidth Extension for Speech.6.1 Applications.6.2 From a Speech Production Model to the Bandwidth Extension Algorithm.6.2.1 Model of the Process of Speech Production.6.2.2 Bandwidth Extension Algorithm.6.2.3 Alternative Structures.6.3 Extension of the Excitation Signal.6.3.1 Explicit Signal Generation.6.3.2 Non-linear Processing.6.3.3 Modulation in the Time Domain.6.3.4 Pitch Scaling.6.3.5 Discussion.6.4 Estimation of the Wideband Spectral Envelope.6.4.1 Representations of the Estimated Spectral Envelope.6.4.2 Instrumental Performance Measure.6.4.3 Theoretical Performance Bound.6.5 Feature Selection.6.5.1 Mutual Information.6.5.2 Separability.6.5.3 Linear Discriminant Analysis.6.5.4 Primary Features.6.5.5 Evaluation.6.6 Codebook Mapping.6.6.1 Vector Quantization and Training of the Primary Codebook.6.6.2 Training of the Shadow Codebook.6.7 Linear Mapping.6.7.1 Training Procedure.6.7.2 Piecewise-linear Mapping.6.8 Gaussian Mixture Model.6.8.1 Minimum Mean Square Error Estimation.6.8.2 Training by the Expectation-maximization Algorithm.6.9 Hidden Markov Model.6.9.1 Statistical Model of the Markov States.6.9.2 Estimation Rules.6.10 Discussion.7 Noise Abatement.7.1 A Special Kind of Noise Reduction.7.2 The Noise Pollution Problem - Case Study.7.3 The Application Low-frequency Psychoacoustic Bandwidth Extension to Noise Pollution.8 Bandwidth Extension Patent Overview.Appendix A Multidimensional Scaling.A.1 Introduction.A.2 Scaling.A.3 Example.A.4 Procedure.A.5 Precautions Concerning the Solution.A.6 Significance of Stress.A.7 Univariate Scaling.References.Index.
TL;DR: A variation of HFCC called HFCC-E is introduced, a modification of MFCC that uses the known relationship between center frequency and critical bandwidth from human psychoacoustics to decouple filter bandwidth from filter spacing, in order to investigate the effects of wider filter bandwidth on noise robustness.
Abstract: Mel frequency cepstral coefficients (MFCC) are the most widely used speech features in automatic speech recognition systems, primarily because the coefficients fit well with the assumptions used in hidden Markov models and because of the superior noise robustness of MFCC over alternative feature sets such as linear prediction-based coefficients The authors have recently introduced human factor cepstral coefficients (HFCC), a modification of MFCC that uses the known relationship between center frequency and critical bandwidth from human psychoacoustics to decouple filter bandwidth from filter spacing In this work, the authors introduce a variation of HFCC called HFCC-E in which filter bandwidth is linearly scaled in order to investigate the effects of wider filter bandwidth on noise robustness Experimental results show an increase in signal-to-noise ratio of 7 dB over traditional MFCC algorithms when filter bandwidth increases in HFCC-E An important attribute of both HFCC and HFCC-E is that the algorithms only differ from MFCC in the filter bank coefficients: increased noise robustness using wider filters is achieved with no additional computational cost
TL;DR: In this paper, the behavior mechanism of the stacked antenna is synthetically clarified for the first time by investigating the results calculated using the FDTD method, and the wide bandwidth and the gain enhancement are considered synthetically by the detailed investigation of calculated results including the near-field distributions.
Abstract: The behaviour mechanism of the stacked antenna is synthetically clarified for the first time by investigating the results calculated using the FDTD method. The stacked microstrip antenna has particular characteristics, such as a high gain or a wide bandwidth. When the size of the parasitic patch, is nearly equal to the fed patch and the distance between the fed patch and the parasitic patch is approximately 0.1 wavelength, the bandwidth is increased. When that distance is approximately half a wavelength, the gain enhancement is obtained. The wide bandwidth and the gain enhancement are considered synthetically by the detailed investigation of calculated results including the near-field distributions. It is shown that the wide bandwidth and the gain enhancement are caused by a two-frequency resonance and leaky resonant cavity formation, respectively. The calculated input impedance and radiation patterns agree well with the experimental values.
TL;DR: In this article, the authors address the critical problem of global wire optimization for nanometer scale very large scale integration technologies, and elucidates the impact of such optimization on power dissipation, bandwidth, and performance.
Abstract: This paper addresses the critical problem of global wire optimization for nanometer scale very large scale integration technologies, and elucidates the impact of such optimization on power dissipation, bandwidth, and performance. Specifically, this paper introduces a novel methodology for optimizing global interconnect width, which maximizes a novel figure of merit (FOM) that is a user-defined function of bandwidth per unit width of chip edge and latency. This methodology is used to develop analytical expressions for optimum interconnect widths for typical FOMs for two extreme scenarios regarding line spacing: 1) spacing kept constant at its minimum value and 2) spacing kept the same as line width. These expressions have been used to compute the optimal global interconnect width and quantify the effect of increasing the line width on various performance metrics such as delay per unit length, total repeater area and power dissipation, and bandwidth for various International Technology Roadmap for Semiconductors technology nodes.
TL;DR: In this article, the relationship between the control bandwidth and the load transient response in voltage regulator modules (VRMs), which are designed with multiphase interleaving synchronous buck converters, is investigated.
Abstract: This paper investigates the relationship between the control bandwidth and the load transient response in voltage regulator modules (VRMs), which are designed with multiphase interleaving synchronous buck converters. Both voltage- and current-mode controls are discussed. A critical bandwidth value is discovered, beyond which pushing the bandwidth can no longer reduce the output voltage spike during the load transient response. Also, the critical bandwidths are different according to different kinds of output capacitors. The critical bandwidth concept highlights the trend of high-frequency VRM design that uses ceramic capacitors to achieve smaller size and faster load transient response. Simulation and experimental results prove the theoretical analysis.
TL;DR: In this paper, a motion video signal encoder maximizes image quality without exceeding transmission bandwidth available to carry the encoded motion video signals by comparing encoded frames of the video signal to a desired size of frame.
Abstract: A motion video signal encoder maximizes image quality without exceeding transmission bandwidth available to carry the encoded motion video signal by comparing encoded frames of the motion video signal to a desired size of frame. If the size of encoded frames differ from the desired size, encoding is adjusted to produce encoded frames closer in size to the desired size. In addition, a cumulative bandwidth error records an accumulated amount of available bandwidth. The cumulative bandwidth error is adjusted as time elapses to add to the available bandwidth and as each frame is encoded to thereby consume bandwidth. As the cumulative bandwidth error grows in magnitude above or below zero, encoding is adjusted as needed to either improve image quality to more completely consume available bandwidth or to reduce image quality to thereby consume less bandwidth and to thereby cause the cumulative bandwidth error to move toward zero. Rapid changes in the amount of change or motion in the motion video signal are detected by comparing the amount of change between two consecutive frames and filtering the amount of change with previously measured amounts of change. Encoding is pre-compensated according to the filtered measurement of rapid change.
TL;DR: In this article, a wideband cable modem system increases available bandwidth of a single channel by encoding a data stream into wideband packets, associated with a logical wideband channel that extends over multiple physical downstream cable channels.
Abstract: A wideband cable modem system increases available bandwidth of a single channel by encoding a data stream into wideband packets. The wideband packets are associated with a logical wideband channel that extends over multiple physical downstream cable channels.
TL;DR: This paper presents an alternative solution where the filter and programmable gain functionality is integrated into a /spl Sigma//spl Delta/ ADC, which becomes highly immune to interferers even if they exceed the maximum allowable input level for the wanted channel.
Abstract: Receivers are being digitized in a quest for flexibility. Analog filters and programmable gain stages are being exchanged for digital processing at the price of a very challenging ADC. This paper presents an alternative solution where the filter and programmable gain functionality is integrated into a /spl Sigma//spl Delta/ ADC. The novel filtering ADC is realized by adding a high-pass feedback path to a conventional /spl Sigma//spl Delta/ ADC while a compensating low-pass filter in the forward path maintains stability. As such, the ADC becomes highly immune to interferers even if they exceed the maximum allowable input level for the wanted channel. As a consequence, the ADC input range can be programmed dynamically to the level of the wanted signal only. This results in an input-referred dynamic range of 89 dB in 1-MHz bandwidth and an intentionally moderate output signal-to-noise-and-distortion ratio of 46-59 dB (depending on the programmed gain). The merged functionality enables a better overall power/performance balance for the receiver baseband. The design consumes less than 2 mW and active area is 0.14 mm/sup 2/ in a 0.18-/spl mu/m digital CMOS technology.
TL;DR: In this article, the authors propose a method for dynamically adjusting the bit rate of the encoder according to a determined available bandwidth for wirelessly communicating source data, where the first encoder bit rate is determined according to the available bandwidth, and the second bit rate can be dynamically adjusted based on the bandwidth available.
Abstract: An apparatus, system and method for use in for use in dynamically adjusting an encoder bit rate according to a determined available bandwidth for wirelessly communicating source data. A method detects a first available bandwidth, determines a first encoder bit rate according to the available bandwidth, encodes a signal at the first encoder bit rate, detects a change in the available bandwidth such that there is a second available bandwidth, determines a second encoder bit rate according to the second available bandwidth, and encodes the signal at the second encoder bit rate. Encoding the signal at the second encoder bit rate can include encoding a subsequent frame of the signal. The method can further wirelessly communicate the signal encoded at the first encoder bit rate in real-time and wirelessly communicate the signal encoded at the second encoder bit rate in real-time.
TL;DR: In this paper, a method of controlling a generator system connected to an electric power system in which the output frequency characteristic of the generator system is measured, a first phase angle and frequency of the measured frequency characteristic is estimated using a first-phase locked loop having a first bandwidth greater than the first bandwidth, and a second-phase angles and frequencies of the estimated frequency characteristic are estimated using an additional loop with a second bandwidth larger than the second bandwidth.
Abstract: A method of controlling a generator system connected to an electric power system in which the output frequency characteristic of the generator system is measured, a first phase angle and frequency of the measured frequency characteristic is estimated using a first phase locked loop having a first bandwidth, and a second phase angle and frequency of the measured frequency characteristic is estimated using a second phase locked loop having a second bandwidth greater than the first bandwidth. Further, the method calculates a frequency difference between the first and second estimated frequencies, and an angle variation that is proportional to the calculated frequency difference. The estimated second phase angle is then added to the calculated angle variation so as to form an output current phase angle reference, and an output current phase angle of the generator system is controlled to be aligned with the output current phase angle reference. The method also determines whether or not the generator system is within a generation island based on the measured frequency characteristic.
TL;DR: In this paper, a system for communicating content streams (i.e., voice, video, data) between a content server (12) connected to a power-line communications network (14) and a remote electronic device (40) having ultra-wide band connectivity with an interface module (16) is presented.
Abstract: A system for communicating content streams (i.e. voice, video, data) between a content server (12) connected to a power-line communications network (14) and a remote electronic device (40) having ultra-wide band connectivity with an interface module (16). The interface module (16) is a power-line communication/ultra-wide band (PLC/UWB) module preferably connected to a conventional power outlet (36) upon the power-line network. The interface module (16) converts the power-line signals from the server (12) to ultra-wide band signals which are broadcast for reception by the wireless electronic device (40). Return communications with the server (12) from the wireless electronic device (40) are also preferably supported. The power-line communication and/or ultra-wide band signal can also be encrypted for improved security. The content server (12) and interface module (16) are preferably configured to communicate content streams only within the portions of the bandwidth allocated by a bus master, which may comprise a content server (12) with programming for controlling bandwidth use on the physical power-line communications network.
TL;DR: In this paper, a thorough analysis of the power delivery path is presented, based on which the current slew rate of each loop is derived, and the relationship between inductor current slew rates of the voltage regulator and the bandwidth is also derived.
Abstract: This paper offers a thorough analysis of the power delivery path. Based on the power delivery path model, the current slew rate of each loop is derived. The relationship between the inductor current slew rate of the voltage regulator (VR) and the bandwidth is also derived. Then, the level of the voltage spike across the capacitors of each loop is determined, after which the relationship between the bandwidth and the capacitance can be plotted. We find that for today's power delivery structure, the bulk capacitors can be eliminated as long as the bandwidth is pushed beyond 350 kHz. The experimental results of a 2-MHz two-stage 12-V VR verify this analysis.
TL;DR: The application of genetic algorithm (GA) optimization to the design and analysis of planar monopole antennas is presented and it is indicated that the RBT can achieve a significantly wider bandwidth with a much smaller size than the traditional BT.
Abstract: The application of genetic algorithm (GA) optimization to the design and analysis of planar monopole antennas is presented. GA is first used to optimize the impedance matching bandwidth of two particular planar element shapes, the bow-tie (BT) and reverse bow-tie (RBT). The results of this study indicate that the RBT can achieve a significantly wider bandwidth with a much smaller size than the traditional BT. In a follow-on study, GA is used to generate arbitrarily shaped planar monopole designs, which exhibit improved broadband performance and/or reduced size compared with the RBT. The designs generated by the GA demonstrate a better tradeoff between matching bandwidth and electrical size compared with planar monopole designs previously characterized in the literature. Analysis of results from simulation and measurement are presented, which provide insight into the operation of these antennas as well as the key parameters that lead to improved performance. Finally, a performance bound is generated to relate the bandwidth limitation of planar monopoles to size.