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Quantum key distribution (QKD) is one quantum technology that can provide secure encryption keys for data transmission. The secret key rate (SKR) is a core performance indicator in QKD, which directly determines the transmission rate of enciphered data. Here, for the first time, we demonstrate a high-key-rate Gaussian-modulated continuous-variable QKD (CV-QKD) using telecom optical components. The framework of CV-QKD over these components is constructed. Specifically, the high-rate low-noise Gaussian modulation of coherent states is realized by a classical optical IQ modulator. High-baud low-intensity quantum signals are received by an integrated coherent receiver under the shot-noise limit. A series of digital signal processing algorithms are proposed to achieve accurate signal recovery and key distillation. The system can yield a high asymptotic SKR of 10.37 Mbps within 20 km standard telecom fiber, and the secure distance can exceed 100 km. This result confirms the feasibility of CV-QKD with state-of-the-art performance using telecom optical components. Besides, due to the ease of integrating these discrete components, it provides a high-performance and miniaturized QKD solution for the metropolitan quantum network. [ABSTRACT FROM AUTHOR]

There is no question about the value that digital signal processing brings to the area of biomedical research. DSP processors are used to sample and process the analog inputs that are received from a human organ. These inputs come from the organ itself. DSP processors, because of their multidimensional data processing nature, are the electrical components that take up the greatest space and use the most power. In this age of digital technology and electronic gizmos, portable biomedical devices represent an essential step forward in technological advancement. Electrocardiogram (ECG) units are among the most common types of biomedical equipment, and their functions are absolutely necessary to the process of saving human life. In the latter part of the 1990s, portable electrocardiogram (ECG) devices began to appear on the market, and research into their signal processing and electronics design capabilities continues today. System-on-chip (SoC) design refers to the process through which the separate computing components of a DSP unit are combined onto a single chip in order to achieve greater power and space efficiency. In the design of biomedical DSP devices, this body of research presents a number of different solutions for reducing power consumption and space requirements. Using serial or parallel data buses, which are often the region that consumes the most power, it is possible to send data between the system-on-chip (SoC) and other components. To cut down on the number of needless switching operations that take place during data transmission, a hybrid solution that makes use of the shift invert bus encoding scheme has been developed. Using a phase-encoded shift invert bus encoding approach, which embeds the two-bit indication lines into a single-bit encoded line, is one way to solve the issue of having two distinct indicator bits. This method reduces the problem. The PESHINV approach is compared to the SHINV method that already exists, and the comparison reveals that the suggested PESHINV method reduces the total power consumption of the encoding circuit by around 30 percent. The computing unit of the DSP processor is the target of further optimization efforts. Virtually, all signal processing methods need memory and multiplier circuits to function properly. [ABSTRACT FROM AUTHOR]

An autonomous surface vehicle known as a wave glider, instrumented with a low-power towed hydrophone array and embedded digital signal processor, is demonstrated as a viable low-noise system for the passive acoustic monitoring of marine mammals. Other key design elements include high spatial resolution beamforming on a 32-channel towed hydrophone array, deep array deployment depth, vertical motion isolation, and bandwidth-efficient real-time acoustic data transmission. Using at-sea data collected during a simultaneous deployment of three wave glider-based acoustic detection systems near Stellwagen Bank National Marine Sanctuary in September 2019, the capability of a low-frequency towed hydrophone array to spatially reject noise and to resolve baleen whale vocalizations from anthropogenic acoustic clutter is demonstrated. In particular, mean measured array gain of 15.3 dB at the aperture design frequency results in a post-beamformer signal-to-noise ratio that significantly exceeds that of a single hydrophone. Further, it is shown that with overlapping detections on multiple collaborating systems, precise localization of vocalizing individuals is achievable at long ranges. Last, model predictions showing a 4× detection range, or 16× area coverage, advantage of a 32-channel towed array over a single hydrophone against the North Atlantic right whale upcall are presented for the continental shelf environment south of Martha's Vineyard. [ABSTRACT FROM AUTHOR]

A novel technique for transmission of information through visible light communication (VLC) is developed in this study. A light-emitting diode is used as the light source at the transmitting side to send the encrypted information. At the receiving side, a light sensor, OPT-101, is used to receive the light signals that carry the encrypted information. The Arduino Due microcontroller board is used for digital signal processing at both the transmitting and receiving sides. Furthermore, to prevent the transmitted message from being intercepted, two chaotic systems, a master and a slave, with a synchronization controller are designed to obtain the transmitted audio signals. The design enables not only a VLC system with the light transmission path as a straight line (so that data cannot be stolen) but also the encryption of the audio signals with the chaotic system (Rössler system) to enhance data transmission security. The effectiveness of this system is then experimentally verified. [ABSTRACT FROM AUTHOR]

High-speed, low-power analog-to-digital converter (ADC) is a critical element for 5-GBd, 20-Gb/s digital signal processing (DSP)-based coherent optical communication receiver. To satisfy high data transmission rate requirement of optical receiver, a single-core and open-loop flash ADC with a new proposed multiplexer-based encoder is presented in this paper. Compared with conventional encoder topology, the new proposed topology achieves the fastest encoding speed and lowest power consumption. The optimized distortion is achieved by utilizing a leakage current compensation technique and a local negative feedback method in switched-buffer track-and-hold amplifier (THA). Strict synchronization is obtained for clock signals by careful designing of layout in tree-based clock networks. Furthermore, a master–slave comparator incorporated with a preamplifier reduces signal-dependent kickback noise as well as offset voltage. By using master–slave comparators and proposed encoders, the sampling rate is up to 21.12 GS/s. The 4-bit, 20-GS/s flash ADC is realized in 0.13- μ m SiGe BiCMOS technology and it only occupies 1.05 mm × 1.46 mm chip area. With a power consumption of 1.831 W from 4-V supply, the ADC achieves an effective number of bits (ENOB) of 2.5 at 15 GS/s. [ABSTRACT FROM AUTHOR]

Although HEVC, the emerging video coding standard, has doubled the coding performance of its predecessor H.264/AVC, its significantly increased computational complexity imposes great obstacles for HEVC encoders to be employed in real-time applications with embedded processors, such as digital signal processors (DSPs). In this paper, a TI Keystone multicore TMS320C6678 DSP-based highly paralleled low-cost fast HEVC encoding solution is well designed and implemented. First, the overall structure of HEVC encoder with CTU-level parallelism is re-designed to well support the encoding parallelism, with full consideration of the hardware characteristics. Second, a low-delay and low-memory multicore data transmission mechanism is proposed to reduce the latency of data access between internal L2 memory and external DDR3. Third, the encoding bottlenecks, i.e., the most time-consuming encoding modules, are identified and optimized for acceleration with TI powerful C6000 SIMD instructions. Experimental results show that our proposed HEVC encoder on TI TMS320C6678 DSPs can significantly improve the real-time capacity with tolerable performance loss, 0.93 dB performance loss under on average 465.50 times speedup as compared to CPU-based HM reference software, more specifically, which makes it desirable in power-constrained real-time video applications. [ABSTRACT FROM AUTHOR]

This paper proposes systematic code design methods for constructing efficient spectrum shaping codes with the maximum runlength limited constraint k, which are widely used in data storage systems for digital consumer electronics products. Through shaping the spectrum of the input user data sequence, the codes can effectively circumvent the interaction between the data signal and servo signal in high-density data storage systems. In particular, we first propose novel methods to design high-rate k constrained codes in the non-return-to-zero (NRZ) format, which can not only facilitate timing recovery of the storage system, but also avoid error propagation during decoding and reduce the system complexity. We further propose to combine the Guided Scrambling (GS) technique with the k constrained code design methods to construct highly efficient spectrum shaping k constrained codes. Simulation results demonstrate that the designed codes can achieve significant spectrum shaping effect with only around 1% code rate loss and reasonable computational complexity. [ABSTRACT FROM PUBLISHER]

This paper gives an information hiding method for H.264/AVC compressed video stream based on integer transform coefficient and virtual space encoding. In this method, the integer transform coefficients which are selected as carriers are divided into multiple groups. And it only needs to change at most the value of one of the coefficients in each group to embed $$\hbox {Log}_{2}(2N+1)$$ binary bits when each group contains $$N$$ coefficients. Moreover, the required change is simple, which can be achieved by adding 1 or $$-$$ 1 to the coefficient. Since only one coefficient's value is slightly changed for every $$\hbox {Log}_{2}(2N+1)$$ secret bits, this method does less harm to the original encoded video stream caused by the information embedding compared with that by current methods. Furthermore, this method only chooses the macroblock that has rich texture as the candidate carrier macroblock, and it employs the embedding position template to select the candidate $$4\times 4$$ coded blocks (I4-blocks) from the carrier macroblock for watermark bits embedding. In this way, the counteracting steganalysis performance is significantly improved. Experimental results on several test sequences demonstrate that the proposed approach can realize blind extraction and the performance is superior to that of the other current state-of-the-art information hiding methods also based on modulating integer transform coefficients. [ABSTRACT FROM AUTHOR]

In this communication, a new method is proposed to estimate the central direction-of-arrival (DOA) of multiple incoherently distributed noncircular sources from using the received data in two closely spaced uniform linear arrays. Both the received data vector and its conjugate are utilized to estimate the extended cross-correlation (CC) matrix. Singular value decomposition is then applied on the extended CC matrix to obtain the left and right singular vectors. Subsequently, the central DOA is estimated by using the left and right singular vectors associated with the largest singular values. The effectiveness and performance of the technique are accessed through numerical results. [ABSTRACT FROM AUTHOR]

In this paper, an interference channel with a cognitive relay (IFC-CR) is considered to achieve spectrum sharing between a licensed primary user and an unlicensed secondary user. The CR assists both users in relaying their messages to the respective receivers, under the constraint that the performance of the legacy primary user is not degraded. Without requiring any non-causal knowledge, the CR uses a successive interference cancellation to first decode the primary and secondary messages after a transmission phase. A power allocation is then performed to forward a linear weighted combination of the processed signals in the relaying phase. Closed-form expressions of the end-to-end outage probability are derived for both primary and secondary users under the proposed approach. Furthermore, by exploiting the decoded primary and secondary messages in the first phase, we propose the use of dirty paper coding (DPC) at CR to pre-cancel the interference seen at the secondary (or primary) receiver in the second phase, which results in a performance upper bound for the secondary (or primary) user without affecting the other user. Simulation results demonstrate that with a joint consideration of the power control at the secondary transmitter and the power allocation at CR, performance gains can be achieved for both primary and secondary users. [ABSTRACT FROM AUTHOR]

The article discusses the 100-Gigabit per second packet transport system developed by Nippon Telegraph and Telephone Corporation Network Service Systems Laboratories for high network traffic loads. The company's 100G-PTS optical technology is said to deliver coherent capacity through Multi-Protocol Label Switch Transport Profile which permits optical and packet switching. Its digital signal processing system is noted to mitigate fiber noise and polarization mode dispersion issues.

The phenomena of alteration of a digital signal that interfere with optimal performance, are present regardless of the number of filters, or dynamic elements electronic devices of high quality and environmental isolates that are available. Because of this, it is essential to have a proper technical management in managing the changes that may occur to the original signal. To correct and align the tax rates on incoming teams high order multiplexing is the applicability of the principles of positive justification or insertion of bits or voids that regulate this speed read-write. On the other hand, data transmission, we have a phenomenon known as jitter or jitter, even causing a shift in the ideal position of the bit received with reference to their nominal flanks, which leads to an inaccuracy final signal from the original income. No digital system must be presumed as immune to these changes. [ABSTRACT FROM AUTHOR]

We present a methodology for nonlinearity compensation amenable to a wide variety of analog-to-digital converters (ADCs). To that purpose, a postcompensation scheme for a commercial ADC is presented and two compensator models are considered: 1) the memory polynomial (MP) and 2) the modified generalized MP. Since the proposed method does not use any information about the compensated architecture, it can be applied to different ADC designs. Furthermore, we address the measurement and characterization setup of the device under test by making a study of the quality of the signals involved to verify the improvement obtained. The issue of the training sequences required by the proposed compensation method is also addressed in detail. Despite the common use of a single training signal, we propose to use several sinusoids in the bandwidth of interest. With this, it is possible to show that the generalization properties of the estimated postcompensator are greatly enhanced compared with the case of a single sinusoid training sequence. As verified by the measurements, considerable gain in accuracy can be obtained using the proposed methodology. In particular, a 10-dB increment in spurious free dynamic range is obtained using the proposed postcompensators over the complete Nyquist frequency band. [ABSTRACT FROM PUBLISHER]

The potential of Time Interference Alignment is investigated in this work, with particular reference to the attainable degrees of freedom. The K-user interference channel is considered, in which transmitters and receivers are placed randomly in a Euclidean space. A model for long delay networks is introduced and the degrees of freedom for different cases (with and without transmitter delay coordination) are evaluated. It is shown how time interference alignment can provide more degrees of freedom than TDMA when the transmitters jointly coordinate their transmission delay and the number of pairs is K ≥ 5. Closed form expressions are derived for several cases of interest which provide insight and useful predictions. This work is concluded with an investigation of the achievable degrees of freedom for multi-satellite networks, where it is shown that the results obtained under several assumptions do predict accurately the degrees of freedom in a real setting. [ABSTRACT FROM PUBLISHER]

Monitoring heart activity from electrocardiograms (ECG) is crucial to avoid unnecessary fatalities; therefore, detection of QRS complex is fundamental to automated ECG monitoring. Continuous, portable 24/7 ECG monitoring requires wireless technology with constraints on power, bandwidth, area, and resolution. In order to provide continuous remote monitoring of patients and fast transmission of data to medical personnel for instantaneous intervention, we propose a methodology that converts analog inputs into pulses for ultralow power implementation. The signal encoding scheme is the time-based integrate and fire (IF) sampler from which a set of signal descriptors in the pulse domain are proposed. Furthermore, a logical decision rule for QRS detection based on morphological checking is derived. The proposed decision logic depends exclusively on relational and logical operators resulting in ultrafast recognition and can be implemented using combinatorial logic hardware to guarantee power consumption orders of magnitude lower than any microprocessor device. The algorithm was evaluated using the MIT-BIH arrhythmia database and results show that our algorithm performance is comparable to the state-of-the art software-based detection. [ABSTRACT FROM PUBLISHER]

Modern GPUs ( Graphics Processing Units) offer very high computing power at relative low cost. To take advantage of their computing resources and develop efficient implementations is essential to have certain knowledge about the architecture and memory hierarchy. In this paper, we use the FFT ( Fast Fourier Transform) as a benchmark tool to analyze different aspects of GPU architectures, like the influence of the memory access pattern or the impact of the register pressure. The FFT is a good tool for performance analysis because it is used in many digital signal processing applications and has a good balance between computational cost and memory bandwidth requirements. [ABSTRACT FROM AUTHOR]

We discuss the emerging rate-adaptable optical transmission technology and how this new technology may be employed to further reduce the transport network cost to meet ever growing bandwidth demand in the core network. Two different types of transponders are considered: those adjusting either the transported bit rate (i.e., client data rate) or the symbol rate (with a fixed bit rate). We propose a methodology for calculating the (normalized) cost to build out an entire long-haul transport network with several options for bit-rate-adaptable transponders. By using link-length demands from an exemplary distance-diverse network, we demonstrate that time-domain hybrid-QAM-enabled fine-grain rate-adaptable transponders can reduce network cost by more than 20 percent within a traditional, fixed-bandwidth, wavelength-division-multiplexed grid. We also argue that the total transponder expense using symbol-rate-adaptable technology will be greater than when using bit-rate-adaptable technology, as well as requiring more costly flex-grid ROADMs for channel routing. [ABSTRACT FROM PUBLISHER]

A novel dispersive delay structure (DDS) based on a composite right/left-handed (CRLH)–CRLH coupler is proposed and demonstrated by both full-wave and experimental results. The experimental prototypes are compact and shielded multilayer DDSs implemented in low-temperature co-fired ceramics technology. Compared to the conventional all-pass C-section DDS, the proposed CRLH C-section DDS exhibit a larger group-delay swing, which leads to higher resolution in analog signal processing applications, as a result of their higher coupling capability. Moreover, they feature a larger bandwidth and a smaller size. A generalized wave-interference analysis is presented to both rigorously derive the transfer functions and group-delay characteristics of C-section DDSs and provide deeper insight into their operating mechanisms in both their left- and right-handed regimes. [ABSTRACT FROM AUTHOR]

This brief presents a simple method to suppress and re-build, if necessary, Internet Protocol and User Datagram Protocol headers for the transmission of signaling metadata when using an all-IP Digital Video Broadcasting (DVB) system. This technique will ease convergence of DVB transmission systems with the Internet. The predicted reduction in packet overhead for signaling when using this technique with the DVB Generic Stream Encapsulation (GSE) is reported. [ABSTRACT FROM PUBLISHER]

We describe the signal processing logic, firmware, and software for a frequency-domain multiplexed biasing and demodulation system that reads out transition-edge-sensor bolometer arrays for millimeter-wavelength cosmology telescopes. This system replaces a mixed-signal readout back end with a much smaller and more power-efficient system relying on field-programmable gate arrays for control, computation, and signal processing. The new system is sufficiently robust, automated, and power efficient to be flown on stratospheric balloon-borne telescopes and is being developed further for satellite applications. [ABSTRACT FROM PUBLISHER]

T he advance of mobile electronics technology has produced handheld appliances allowing both wireless voice and data communications. One of the most important operations in the realm of digital signal and image processing is the 2-D Discrete Cosine Transform. This paper presents a multiplierless two dimensional Discrete Cosine Transform/Inverse Discrete Cosine Transform (DCT/IDCT) based on the transpose method. In this method the 2-D DCT is obtained by taking two 1-D DCTs in series. The input data is first divided into NxN blocks and the row-wise 1-D DCT of each block is taken, the intermediate transposition is then determined and a column-wise 1-D DCT is ascertained which gives the 2-D DCT of the data. The hardware implementation is parallel, pipelined and decomposed the coefficients matrix into four power of two term(i.e:16 ) to perform shift and add operations instead of multipliers(i.e 16); it costs only 1,443 slice , and runs at maximum frequency of 82.8 MHz with a very high process throughput of 991.2 Megabits/sec when synthesized onto Spartan3-E XC3S500 FPGA device. The proposed 2-D DCT/IDCT design achieving the most demanding real-time requirements of CODEC standardized frame resolutions and rates. [ABSTRACT FROM AUTHOR]

There exist a variety of ways to represent 3D content, including stereo and multiview video, as well as frame-compatible and depth-based video formats. There are also a number of compression architectures and techniques that have been introduced in recent years. This paper provides an overview of relevant 3D representation and compression formats. It also analyzes some of the merits and drawbacks of these formats considering the application requirements and constraints imposed by different storage and transmission systems. [ABSTRACT FROM AUTHOR]

This paper develops a coordination control strategy for wind farms with line commutated converter-based HVdc delivery for participating in inertial response and primary frequency control. The coordination philosophy is to feedback the grid frequency and its derivative and adjust the delivery power of the HVdc link according to the feedback signals. The feedback loop employing the derivative of the grid frequency aims to improve the inertial response while the feedback loop employing the grid frequency deviation introduces a droop at the rectifier control loop. When the grid frequency is too high or too low, active power flow through the HVdc link will be ramped down or up. In turn, the wind generation will increase or decrease the blade angles to reduce or increase the captured wind power through pitch control. A case study demonstrates the effectiveness of the inertial enhancement and frequency droop in HVdc control. Simulation results in transient security assessment tool are given. [ABSTRACT FROM AUTHOR]

Since the early days of digital communication, hidden Markov models (HMMs) have now been also routinely used in speech recognition, processing of natural languages, images, and in bioinformatics. In an HMM (Xt, Yt)t≥1, observations X1, X2,... are assumed to be conditionally independent given a Markov process Y1, Y2,..., which itself is not observed; moreover, the conditional distribution of Xt depends solely on Yt. Central to the theory and applications of HMM is the Viterbi algorithm to find a maximum a posteriori probability (MAP) estimate v(x1:T) = (v1, v2,..., vT) of Y1:T given observed data x1:T. Maximum a posteriori paths are also known as the Viterbi paths, or alignments. Recently, attempts have been made to study behavior of the Viterbi alignments when T → ∞. Thus, it has been shown that in some cases a well-defined limiting Viterbi alignment exists. While innovative, these attempts have relied on rather strong assumptions and involved proofs which are existential. This work proves the existence of infinite Viterbi alignments in a more constructive manner and for a very general class of HMMs. [ABSTRACT FROM AUTHOR]

This paper describes a simultaneous register and functional unit (FU) binding algorithm in high level synthesis. Our algorithm targets the reduction of multiplexer inputs, shortening the total length of global interconnects. Specifically, our algorithm maximizes the interconnect sharing among FUs and registers by considering flow dependences, common primary inputs, and common register inputs among operations. Experimental results have shown that our scheme achieves more than 20% multiplexer input count reduction, on average, over previously proposed algorithms. Our approach delivers a 18% wirelength reduction of global interconnects with minor area overhead. [ABSTRACT FROM AUTHOR]

Digital signal transmission can be viewed as latticetiling over a time-frequency plane. Corresponding to a rectangular lattice and a hexagonal lattice, there are two multicarrier transmission schemes, i.e., the Weyl-Heisenberg (W-H) system and the hexagonal Gabor (H-G) system, respectively. In the previous works, the transmission pulse shape and the time-frequency lattice parameters of the above two systems were optimized to obtain the best robustness against time-frequency dispersive channels. In this paper, by virtue of elliptic integral theory, we theoretically prove that the optimized W-H system and H-G system are asymptotically equivalent to each other in the sense of achieving the same energy perturbation as signaling efficiency approaches 0 or +8. Moreover, it is shown that, for moderate signaling efficiencies, the superiority of the H-G system over the W-H system is virtually limited. Numerical results are provided to support the theoretical analysis. [ABSTRACT FROM AUTHOR]

One of the most significant types of error in digital signal processing (DSP) systems working with wideband signals is the error introduced by the analog-to-digital (AD) and digital-to-analog (DA) converters. This paper presents an accurate and simple method to evaluate the performance of AD/DA converters affected by clock jitter, which is based on the analysis of the mean square error (MSE) between the reconstructed signal and the original one. Using an approximation of the linear minimum MSE (LMMSE) filter as reconstruction technique, we derive analytic expressions of the MSE. In particular, through asymptotic analysis, we are able to simply evaluate the performance of digital signal reconstruction as a function of the clock jitter, number of quantization bits, signal bandwidth and sampling rate. [ABSTRACT FROM AUTHOR]

Signal processing applications have high instruction level parallelism (ILP) and real-time performance requirements. Embedded and application specific multicluster architecture is desirable to provide the large computation power that these applications need. As technology moves to deep submicron level, it becomes more important and challenging to design an efficient intercluster connection network to satisfy the rapid growing intercluster data transfer needs under the power and cost constraints. This paper addresses the automatic generation of intercluster connection network with partially connected buses. An application specific approach is proposed in this paper to determine the minimum number of required partially connected buses without performance degradation for a given schedule in polynomial time. The intercluster connection topology is then generated with the determined minimum number of partially connected buses to minimize the connection bus segments. Further, a scheduling algorithm is presented in this paper to minimize the intercluster communication needs for the given application and to reduce the minimum number of partially connected buses required in the interciuster connection network under schedule length constraint. Experimental results indicate that an average reduction up to 50.6% in the number of minimum required buses and an average reduction of 64.5% in bus segments can be achieved compared to commonly used intercluster communication aware scheduling techniques and as soon as possible (ASAP) data transfer scheme. [ABSTRACT FROM AUTHOR]

A compact uniplanar design of 2-D antenna arrays is presented for digital beamforming (DBF) and direction-of-arrival (DOA) estimations. The array is based on switchable microstrip slot antennas that are series fed in one dimension and parallel fed in the other dimension. In conjunction with spatial multiplexing of local elements (SMILE) techniques, the array is able to use both the series- and parallel-fed antennas for 2-D beamforming and DOA estimations. The combination of series and parallel feeding provides a simple way of constructing 2-D DBF arrays with a simple and uniplanar feed network. Other advantages are its low cost and compatibility with a microwave integrated circuit process for the receiver system. A 4 x 4 receiving array test bed at a 5.5-GHz operating frequency is fabricated, and DBF and DOA tests on this test bed are successfully demonstrated. [ABSTRACT FROM AUTHOR]

For a linear controllable system, the problem of reconstruction of all input signals, which are compatible with the measured signal, is considered. It is assumed that information can be transmitted in a processing center only via digital communication channel at discrete time instants, and the word length is limited. In this connection, there are encoding and decoding devices in the communication channel. For simplicity, the communication channel is assumed to be noiseless and delay-free. Defining relationships are obtained for the set of compatible input signals, as well as relationships between reconstruction precision, length of transmitted word and transmission frequency. [ABSTRACT FROM AUTHOR]

In a GNSS software receiver, blocks of digitized intermediate-frequency (IF) data are prepared at the front end and sent to the processing unit for signal acquisition, tracking, demodulation, and position fix. A problem that is observed in practice when the GNSS software module is co-located with a mobile phone device is that the digitized IF data are subject to data intermittency. The intermittency results from the use of mobile phone when the GNSS signals are being received or the execution of higher priority routine in the host computer. An improper buffering scheme at the front end may also lead to such phenomena. Such data intermittency leads to bursttype errors (rather than bit-type errors) on the data to be processed, posing a challenge for the tracking of the GNSS signals. Indeed, if not properly handled, a typical tracking algorithm is not able to accommodate such a large data gap. In the paper, a decision-aided tracking algorithm is developed for a seamless tracking of codes and phases. The effects of data intermittency on code and phase tracking are analyzed. To accounting for the effect without going through the reacquisition process, a detector based on signal power and continuity is then implemented to facilitate decision making. It is shown, through the processing of field gathered data, that the proposed algorithm can indeed be able to continuously track codes and phases in the presence of data intermittency. [ABSTRACT FROM AUTHOR]

Three hybrid digital-analog (HDA) systems, denoted by HDA-I, HDA* and HDA-II, for the coding of a memory- less discrete-time Gaussian source over a discrete-time additive memoryless Gaussian channel under bandwidth compression are studied. The systems employ simple linear coding in their analog component and superimpose their analog and digital signals before channel transmission. Information-theoretic upper bounds on the asymptotically optimal mean squared error distortion of the systems are obtained under both matched and mismatched channel conditions. Allocation schemes for distributing the channel input power between the analog and the digital signals are also examined. It is shown that systems HDA* and HDA-II can asymptotically achieve the optimal Shannon-limit performance under matched channel conditions. Low-complexity and low- delay versions of systems HDA-I and HDA-II are next designed and implemented without the use of error correcting codes. The parameters of these HDA systems, which employ vector quantization in conjunction with binary phase-shift keying modulation in their digital part, are optimized via an iterative algorithm similar to the design algorithm for channel-optimized vector quantizers. Both systems have low complexity and low delay, and guarantee graceful performance improvements for high CSNRs. For memoryless Gaussian sources the designed HDA- II system is shown to be superior to the HDA-I designed system. When applied to a Gauss-Markov source under Karhunen-Loeve processing, the HDA-I system is shown to provide considerably better performance. [ABSTRACT FROM AUTHOR]

Hybrid digital-analog coding schemes have been proposed in source-channel coding to increase the robustness toward channel mismatch, in the absence of transmitter channel state information (CSIT). Recognizing that the same kind of robustness is needed at the relay in a three-node relay network, we propose several novel relaying protocols based on hybrid digital-analog transmission. We compare the performance of the new schemes with traditional digital-only (decode-and-forward or compress-and-forward) or analog-only (amplify-and-forward) relaying, as well as to performance bounds corresponding to genie-aided compress-and-forward relaying. Our new protocols achieve significant gains in terms of achievable expected rates, and they are able to close in on the performance bounds. In particular, we conclude that the best overall performance is obtained by an adaptive combination of decode-and-forward and hybrid digital-analog relaying. [ABSTRACT FROM AUTHOR]

This paper presents the use of the common-mode signal in digital-subscriber-line (DSL) transmission. The DSL binder's metallic sheath is used as a common reference to obtain reliable channels for the common-mode signals. The direct and crosstalk channels for the common-mode signal are obtained using multi-conductor transmission line theory, and the data-rate improvement by exploiting these extra dimensions is presented. The lines in the binder are coordinated to cancel differential-mode crosstalk and also to avoid interference caused by radiation from the common-mode within the binder. The high level of common-mode noise is countered by exploiting the correlation between the differential- and common-mode noises, Numerical results based on the derived models show data-rate improvements of up to 100% compared to using only differential-mode transmission. [ABSTRACT FROM AUTHOR]

A random-access-based feedback protocol with a reservation-based (RB) channel is proposed for multiuser diversity in a wireless time-division-duplex system. The feedback channels consist of one RB channel and several contention-based (CB) channels. The users randomly access the CB channels if their channel states exceed a threshold, and a base station (BS) schedules a user who reports the highest signal-to-noise ratio from multiple users that successfully provide feedback. If there is no successful feedback in the CB channels, the BS uses the feedback information that a designated user reports through the RB channel. The proposed feedback protocol achieves an almost ideal sum-rate capacity with a fixed number of feedback channels regardless of the number of users. [ABSTRACT FROM AUTHOR]

Orthogonal frequency-division multiplexing (OFDM) can be used to support high-data-rate transmissions over time-dispersive fading channels. Many OFDM-based communication systems use packet-based transmission, where channel estimation is needed for the detection of information-carrying symbols. The finite impulse response (FIR) channel model is simple and effective for some simulations of channel estimation for the OFDM-based communication systems over the time-dispersive channels; yet, it is only an approximate channel model that cannot be used in the case of accurate channel estimation. Unfortunately, many researchers have overlooked this issue and have been devising channel-estimation algorithms squarely based on the FIR channel model. While the channel estimation results from these algorithms can be impressive for the FIR channels, the algorithms can hardly be applied in real-world applications. This paper explains in detail the reason the FIR channel model is only an approximate channel model for the OFDM-based communication systems, trying to discourage the inappropriate usage of this model, which can lead to fruitless efforts. This paper also presents a realistic channel model for OFDM-based communication systems, which can be used to realistically access the channel parameter-estimation algorithms. [ABSTRACT FROM AUTHOR]

A new method is proposed to construct space-frequency (SF) codes for multiple-input-multiple-output (MIMO) orthogonal frequency-division multiplexing (OFDM) systems undergoing frequency-selective fading. The new design exploits the geometric relationship of coherent and noncoherent SF codes, and each codeword is constructed as a product of a noncoherent SF codeword and an information-bearing square matrix. A two-stage decoding strategy, where noncoherent detection and coherent detection are implemented sequentially, is proposed to reduce decoding complexity. [ABSTRACT FROM AUTHOR]

Due to the well-separated lattice structure, the excellent time-frequency (T-F) concentration of the modulation pulses, and the underspread property of the mobile radio channels, the pulse cross-correlation matrix (PCCM) at the receiver in a properly designed hexagonal multicarrier transmission scheme essentially takes on a band-limited and sparse structure. By neglecting the almost-zero-valued entries in the PCCM and rearranging the data vector with the aid of the pilot symbols, we propose a parallel maximum-likelihood sequence detection (MLSD), which decomposes the original large-sized MLSD into several small-sized MLSDs that independently operate. With the parallel MLSD via the dynamic programming algorithm, the global maximization of the log likelihood is guaranteed. The signal-to-noise ratio (SNR) loss due to such idealization of the PCCM is analytically evaluated, It is shown that, when the signaling efficiency, i.e., a quantity characterizing the density of symbols in the T-F plane, is not so large, the performance degradation is negligible, compared with that caused by the background noise in practical environment. Simulation results are presented to confirm the effectiveness of the proposed parallel MLSD scheme. [ABSTRACT FROM AUTHOR]

We consider a combined time-hopping (TH) and spread-time (ST) multiple-access technique that uses an internal code. In this method, the duration of each bit is divided into N8 frames. The outputs of the encoder and a pseudorandom (PN) sequence specify the number of the frame in which the data bit is transmitted in ST code-division multiple-access (ST-CDMA) form using the second PN sequence. We consider the correlator receiver, followed by the channel decoder. We obtain the performance of the combined method in additive white Gaussian noise (AWGN) and fading channels in the presence of multiple-access interference (MAI) and narrowband interference (NBI). We also consider the conventional ST-CDMA system and compare its performance with our method. The results indicate that the proposed method has much better performance while it has the same bandwidth and spectral efficiency as the conventional ST-CDMA system. In the new method, like in the conventional ST-CDMA system, an external error-correcting code can be applied for further performance enhancement. [ABSTRACT FROM AUTHOR]

In this paper, we present the results of the measurement and modeling of ultrawideband (UWB) time of arrival (TOA)-based ranging in different indoor multipath environments. We provide a detailed characterization of the spatial behavior of ranging, where we focus on the statistics of the ranging error in the presence and absence of the direct path (DP) and evaluate the path loss behavior in the former case, which is important for indoor geolocation coverage characterization. The frequency-domain measurements were conducted, with a nominal frequency of 4.5 GHz with two different bandwidths, i.e., 500 MHz and 3 GHz. The parameters of the ranging error probability distributions and path loss models are provided for different environments (e.g., an old office, a modern office, a house, and a manufacturing floor) and different ranging scenarios [e.g., indoor to indoor (ITI), outdoor to indoor (OTI), and roof to indoor (RTI)]. [ABSTRACT FROM AUTHOR]

A novel and simple semianalytical method for evaluating the average probability of transmission error for digital communication systems that operate over slow-fading channels is presented. The proposed method applies a sum of exponentials fit known as the Prony approximation to the conditional probability of error. Hence, knowledge of the moment-generating function of the instantaneous signal-to-noise ratio (SNR) at the detector input can be used to obtain the average probability of error. Numerical results show that knowledge of the conditional probability of error at only a small number of points and the sum of only two exponentials are sufficient to achieve very high accuracy; the relative approximation error of the exact average probability of error is less than 6% in most of the cases considered. Furthermore, a piecewise polynomial approximation of the conditional probability of error is investigated as an alternative to the sum of exponentials fit. In this case, knowledge of the partial moments of the instantaneous SNR at the detector input can be used to obtain the average probability of error. Numerical results indicate that, to achieve good accuracy, the method based on the polynomial approximation requires that the product of the polynomial degree and the number of approximation subintervals be larger than 10. [ABSTRACT FROM AUTHOR]

The bit-error-rate (BER) estimation of communication systems can be very time consuming when dealing with complex communication systems and complex modulation schemes such as orthogonal frequency-division multiplexing (OFDM). The signals of such modulation schemes have crest factors (also known as peak-to-average power ratios) that strongly depend on the transmitted symbol. This paper describes a quasi-analytical method to compute the BER, which takes the probability density function (pdf) of the crest factor into account. For each crest factor, the BER is computed using either a Monte Carlo estimate or a quasi-analytical method, which uses the Best Linear Approximation (BLA) model to determine the signal-to-distortion ratio. This BLA model determines a best linear model (in least-squares sense) of the nonlinear system, together with additive noise sources that comprise not only additive noise but also stochastic nonlinear distortions. This best linear model also includes the compression characteristics introduced by the nonlinear system. The proposed quasi-analytical BER method is demonstrated on measurements of an asymmetric digital subscriber line (ADSL) modem driver and on system-level simulations of an OFDM wireless local area network (WLAN) system. The results show that the technique achieves similar accuracies with almost two orders of magnitude fewer simulations than without taking the pdf of the crest factor into account. [ABSTRACT FROM AUTHOR]

The problem of source-channel coding over a multiple-antenna (MIMO) channel with quantized channel state information at the transmitter (CSIT) is considered. Upper bounds on the distortion exponents achieved with partial CSIT under a long-term power constraint are developed. It is shown that the distortion exponent with perfect CSIT grows unbounded as the ratio between the channel and source bandwidth increases, while the exponent achieved with any feedback link of fixed, finite resolution is bounded above by a polynomial of the product between the number of transmit and number of receive antennas. The resolution of the feedback link should grow with the bandwidth ratio to make the distortion exponent scale as fast as that in the perfect- CSIT case. We show that in order to achieve the optimal scaling the CSIT feedback resolution must grow logarithmically with the bandwidth ratio for MIMO channels, and faster than linear for the single-input single-output channel. The achievable distortion exponent of some hybrid schemes with heavily quantized feedback is also derived. The results demonstrate that dramatic performance improvement over the case of no CSIT can be achieved by combining simple schemes with a very coarse CSIT feedback. [ABSTRACT FROM AUTHOR]

This paper presents a modelling of a rectifier with unity power factor (PFC) in half bridge configuration. The proposed approach is based on the movable average technique and quasi-static process analysis. This methodology is applied to find the transfer functions that describe the dynamics of the PFC, under a cascade control structure of internal and external loops. This control requires a larger bandwidth for the internal loop. Finally, some experimental results with a digital control implemented in a digital signal processor (DSP) are presented. [ABSTRACT FROM AUTHOR]

Recently, Multi-User Selection Diversity (MUSDiv) for single-carrier systems has been under extensive study on account of the enhancement it provides to system performance with minimum feedback requirements. However, its application to multichannel systems is considered straightforward and thus, it has not been thoroughly examined. In this paper, the performance of MUSDiv is investigated when applied to the spread-spectrum multi-carrier multiple-access system, where the scheduling has to be performed for all the available channels and self interference must also be considered. Specifically, based on the absolute and normalized Signal-to-Noise-Ratio (SNR) scheduling algorithms, two algorithms are presented, modified and optimized, so that they can be applied on a subband instead of a single-channel basis. Moreover, we propose a new scheduling scheme which constitutes a trade-off between the previous schemes, concerning fairness and capacity performance. The new algorithms are related to the symbol Signal-to-Noise-plus-Interference-Ratio (SNIR) instead of SNR and two interference models were devised to this end. Closed-form expressions for the system capacity are extracted for each case, which are compared with simulation results. The research is also extended to the case of non-identical user power profiles among the available subcarriers. The channel state information required to utilize multi-user selection diversity is already necessary for the most common combining schemes, so no more feedback is actually needed. [ABSTRACT FROM AUTHOR]

In this paper we study and analyze the performance of the RAKE receiver for a typical Ultrawideband (UWB) Spectrally-Encoded Spread-Time (ST) CDMA in the presence of multiple Gaussian Interference (GI) signals. We demonstrate that by combining useful properties of three-level codes, ie., codes with values of -1,0,+1, and RAKE receiver, we can introduce a new strategy in which it can result in superior performance when compared to previous results. In particular, following a discussion on three-level codes we can show that RAKE receiver with optimum multiplicative weights achieves a better performance when compared with simpletwo-level codes. With the help of an example we show various properties of RAKE receiver as well as three-level codes and relate these properties to a number of simple descritive parameters of the communication media. Simulation results obtained in this paper indicate the precision with which the analytical results have been obtained in modeling the above system. [ABSTRACT FROM AUTHOR]

Transmit beamforming is an energy-efficient wireless communication technique that allows a transmitter with two or more antennas to focus its bandpass signal in an intended direction. Recently, the idea of transmit beamforming has been extended to networks of single-antenna cooperative transmitters that pool their antenna resources and behave as a "distributed beamformer." Unlike conventional transmit beamforming, however, the carriers of the single-antenna transmitters in a distributed beamformer are each synthesized from independent and imperfect local oscillators; carrier phase and frequency synchronization among the transmitters is necessary to ensure that a beam is aimed in the desired direction. In this paper, a new time-slotted round-trip carrier synchronization protocol is proposed to enable distributed beamforming in multiuser wireless communication systems. Numerical results for a distributed beamformer with two transmitters show that the parameters of the round-trip synchronization protocol can be selected such that a desired level of phase accuracy and reliability is achieved during beamforming and such that the synchronization overhead is small with respect to the amount of time that reliable beamforming is achieved. The impact of mobility on the performance of the round-trip carrier synchronization protocol is also shown to be small when the synchronization timeslots are short. [ABSTRACT FROM AUTHOR]

High peak-to-average-power ratio (PAPR) is a particular disadvantage of orthogonal frequency-division multiplexing (OFDM) systems and can significantly degrade the power efficiency at the transmitter. The aim of this paper is to propose a post-inverse fast Fourier transform (IFFT) PAPR reduction method based on intelligent linear symbol combining with the objective to minimize the PAPR at the transmitter. Unlike the methods so far reported in the literature, the proposed method has a relatively low complexity and does not require external randomization sequences. This paper will also investigate the impact of the proposed method on the system's bit error rate (BER) performance and provide a closed-form analytical expression that takes into account the BER dependency on the side information bits and multiblock combination. It will be shown that the proposed technique can provide significant PAPR reductions at manageable levels of signal processing requirements while the BER degradation is relatively small. [ABSTRACT FROM AUTHOR]

Scalable representation of visual signals, such as image and video signals, has become a subject of active research since early 1980s. Scalability allows the adaptation of the bit rate and/or the resolution of the transmitted data to the network bandwidth and/or the rendering capability of the receiving device. For many years, spatial scalability has been achieved through wavelets, but recently the Laplacian pyramid (LP) has become an alternative choice because of reduced aliasing in the lower resolutions. In this paper, we focus on the coding efficiency of the LP with a view to transmitting it over a communication channel. In particular, we aim to improve the compression efficiency of the LP detail layers through improved interlayer prediction and orthogonal spatial transforms. First, we consider an LP in the open-loop configuration and propose to improve its rate-distortion performance by compressing it to a critically sampled representation. We derive four different orthogonal spatial transforms from the upsampling and downsampling filters that can achieve this representation, and apply them on the detail layers. The application of these transforms to the detail layers renders a fixed number of transform coefficients either zero or redundant, thus making their transmission unnecessary. Then we consider the compression of an LP in the closed-loop configuration through similar spatial transforms. Because of the introduction of quantization in the prediction loop, these spatial transforms applied on the detail layers do not produce the same number of zero or redundant transform coefficients as in the open-loop case. Nevertheless, the insight obtained from the open-loop coding leads us to enhance the interlayer prediction, and the subsequent application of the spatial transforms to the new detail layers aims to achieve better energy compaction. [ABSTRACT FROM AUTHOR]