Search

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 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]

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]

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]

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]

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]

We propose and evaluate a novel method of constructing Hybrid Automatic Repeat reQuest (HARQ) systems using the specific properties of the Bahl, Cocke, Jelinek, and Raviv (BCJR) error-correcting algorithm. Because the convergence to the actual codeword is not always guaranteed with the BCJR approach, we propose a system in which two different types of Negative AcKnowledgement messages (NAKs) are employed. The first type is the conventional 1-bit NAK, and the second type specifies retransmission pattern in such a way that the additional parity bits are concentrated on the parts of the code trellis that did not converge to a valid sequence. This is different from the traditional construction of rate-compatible punctured codes (RCPCs), which attempts to obtain the optimal weight distance properties of the codes without taking the convergence properties of the BCJR decoder into account. We demonstrate the performance of the algorithm using RCPCs, and we show that our system outperforms the best known conventional HARQ scheme in terms of the throughput and the average length of retransmitted blocks on practical Gaussian, Rayleigh, and thresholded Rayleigh channels. Moreover, as opposed to other adaptive HARQ algorithms, our solution requires no precomputed lookup tables, and it is robust to changes in the channel characteristics and only introduces moderate increase in feedback link throughput requirements. [ABSTRACT FROM AUTHOR]

In this paper, we consider blind channel estimation for cyclic-prefixed single-carrier (CP-SC) systems via second-order statistics only. By exploiting the existence of real symbols in the transmitted data block, we develop a simple technique for blind channel estimation that does not need redundant precoding or virtual carriers. Hence, the proposed algorithm provides band-width efficiency. A side benefit of the proposed algorithm is that the phase ambiguity, which is an inherent problem in traditional blind channel estimation, is reduced to sign ambiguity only. This side benefit facilitates signal detection in digital communications where symbols are usually taken from a finite alphabet. Although the proposed method is mainly developed for channel estimation in single-input-single-output CP-SC systems, it can also be applied straightforwardly to single-input-multiple-output systems (either with CP-SC or other kinds) and multiple-input-multiple-output systems whenever real symbols exist or are deliberately inserted in the transmission. [ABSTRACT FROM AUTHOR]

We report an RSFQ Digital Signal Processor design based on hybrid RSFQ-CMOS memory suitable for a general matrix-on-matrix multiplication algorithm. The DSP consists of an RSFQ Multiply-Accumulate Unit, memory caches and synchronization block, partitioned into multiple chips, and a large CMOS memory. The parameters of the RSFQ DSP are a 10 x 10 bits multiplication with rounding to 14 bits, an 18-bit accumulator length and a 3.7 Kb memory cache. The maximum simulated clock frequency is equal to 24 GHz for HYPRES 4.5 kA/cm2 process and optimum communication bandwidth with the CMOS memory is 2 Gbps. The simplified version of the RSFQ DSP consisting of a 4 x 4 MAC with rounding to 5 bits and 17 x 6 memory caches has been designed for HYPRES 4.5 kA/cm2 process. [ABSTRACT FROM AUTHOR]

In this paper, a new, efficient class of blind equalization algorithms is proposed for use in high-order, two-dimensional digital communication systems. We have called this family: the Constant Norm Algorithms (CNA). This family is derived in the context of Bussgang techniques. Therefore, the resulting algorithms are very simple. We show that some well-known blind algorithms such as "Sato's algorithm" or the Constant Modulus Algorithm (CMA) are particular cases in our CNA family. In addition, from this class, a new cost function, named Constant sQuare Algorithm (CQA), is derived, which is well designed for QAM. It results in a lower algorithm noise without increasing the complexity. Another advantage of this class lies in the possibility of creating new norms by combining several existing norms in order to benefit from the advantages of each original norm. For example, we present the norm resulting from the combination of the two algorithms, CMA and CQA. Moreover, we highlight that, with regard to the excess mean-square error performance, there is an optimal norm for each constellation, i.e., each modulation, in order to equalize it blindly. [ABSTRACT FROM AUTHOR]

CORDIC is a versatile algorithm widely used for VLSI implementation of digital signal processing applications. This article presents a tutorial of how to use CORDIC to implement different communication subsystems that can be found in a software defined radio. Specifically, it shows how to use CORDIC to implement direct digital synthesizers, AM, PM, and FM analog modulators and ASK, PSK and FSK modulators, up-/down-converters of in-phase and quadrature signals, full mixers for complex signals, and phase detection for synchronizers. The article also shows some tricks to efficiently implement the algorithm. [ABSTRACT FROM AUTHOR]

In this paper, we propose a self-synchronization algorithm for audio watermarking to facilitate assured audio data transmission. The synchronization codes are embedded into audio with the informative data, thus the embedded data have the self-synchronization ability. To achieve robustness, we embed the synchronization codes and the hidden informative data into the low frequency. coefficients in DWT (discrete wavelet transform) domain. By exploiting the time-frequency localization characteristics of DWT, the computational load in searching synchronization codes has been dramatically reduced, thus resolving the contending requirements between robustness of hidden data and efficiency of synchronization codes searching. The performance of the proposed scheme in terms of SNR (signal to noise ratio) and BER (bit error rate) is analyzed. An estimation formula that connects SNR with embedding strength has been provided to ensure the transparency of embedded data. BER under Gaussian noise corruption has been estimated to evaluate the performance of the proposed scheme. The experimental results are presented to demonstrate that the embedded data are robust against most common signal processing and attacks, such as Gaussian noise corruption, resampling, requantization, cropping, and MP3 compression. [ABSTRACT FROM AUTHOR]

In this paper, we present a new filter decomposition method for multichannel blind deconvolution of nonminimum-phase systems. With this approach, we decompose a doubly finite impulse response filter into a cascade form of two filters: a causal finite impulse response (FIR) filter and an anticausal FIR filter. After introducing a Lie group to the manifold of FIR filters, we discuss geometric properties of the FIR filter manifold. Using the nonholonomic transform, we derive the natural gradient on the FIR manifold. By simplifying the mutual information rate, we present a very simple cost function for blind deconvolution of nonminimum-phase systems. Subsequently, the natural gradient algorithms are developed both for the causal FIR filter and for the anticausal FIR filter. Simulations are presented to illustrate the validity and favorable learning performance of the proposed algorithms. [ABSTRACT FROM AUTHOR]

Significant power reduction can be achieved by exploiting real-time variation in system characteristics. An approach is proposed and studied herein that exploits variation in signal transmission system characteristics to reduce power consumption while decoding convolutional codes. With this approach, Viterbi decoding is adaptively approximated by varying the pruning threshold of the T-algorithm and truncation length while employing trace-back memory management. A heuristic is given for finding and adaptively applying pairs of pruning threshold and truncation length values that significantly reduce power to variations in signal-to-noise ratio (SNR), code rate, and maximum acceptable bit-error rate (BER). The power reduction potential of different levels of adaptation is studied. High-level energy reduction estimates of 80% to 97% compared with Viterbi decoding are shown. Implementation insight and general conclusions about when applications can particularly benefit from this approach are given. [ABSTRACT FROM AUTHOR]

This paper addresses the problem of blind channel equalization in the context of digital communications. Recent results have shown that certain operations applied to the source signal at the transmitter help in the blind identification and equalization of the channel at the receiver. In this paper, the baseband data signal is multiplied with a chirp sequence. Exploiting certain structural properties arising from this operation, a batch-type algorithm is obtained for calculating the equalizer's coefficients. Conditions on the chirp sequence parameters are obtained that guarantee an equalization solution. A low-complexity adaptive algorithm is also proposed. Finally, extensive simulations, and comparisons with other well-known blind techniques, illustrate the excellent performance of this algorithm. [ABSTRACT FROM AUTHOR]

Multiple on-chip memory modules are attractive to many high-performance digital signal processing (DSP) applications. This architectural feature supports higher memory bandwidth by allowing multiple data memory accesses to be executed in parallel. However, making effective use of multiple memory modules remains difficult. The performance gain in this kind of architecture strongly depends on variable partitioning and scheduling techniques. In this paper, we propose a graph model known as the variable independence graph (VIG) and algorithms to tackle the variable partitioning problem. Our results show that VIG is more effective than interference graph for solving variable partitioning problem. Then, we present a scheduling algorithm known as the rotation scheduling with variable repartition (RSVR) to improve the schedule lengths efficiently on a multiple memory module architecture. This algorithm adjusts the variable partitions during scheduling and generates a compact schedule based on retiming and software pipelining. The experimental results show that the average improvement on schedule lengths is 44.8% by using RSVR with VIG. We also propose a design space exploration algorithm using RSVR to find the minimum number of memory modules and functional units satisfying a schedule length requirement. The algorithm produces more feasible solutions with equal or fewer number of functional units compared with the method using interference graph. [ABSTRACT FROM AUTHOR]

Highlights the development of a scalable VLSI architecture, with a two-channel quadrature mirror filter lattice, based on discrete wavelet transformation, in relation to digital signal processing. In-depth look at the data format converter's stucture; Discussion on data transfer; Characteristics of discrete wavelet transform; Information on discrete wavelet transform scheduling algorithms.

At 2000 m depth in the oceans, one can hear biological, seismological, meteorological, and anthropogenic activity. Acoustic monitoring of the oceans at a global scale and over long periods of time could bring important information for various sciences. The Argo project monitors the physical properties of the oceans with autonomous floats, some of which are also equipped with a hydrophone. These have a limited transmission bandwidth requiring acoustic data to be processed on board. However, developing signal processing algorithms for these instruments requires one to be an expert in embedded software. To reduce the need of such expertise, we have developed a programming language, called MeLa. The language hides several aspects of embedded software with specialized programming concepts. It uses models to compute energy consumption, processor usage, and data transmission costs early during the development of applications; this helps to choose a strategy of data processing that has a minimum impact on performances. Simulations on a computer allow for verifying the performance of the algorithms before their deployment on the instrument. We have implemented a seismic P wave detection and a blue whales D call detection algorithm with the MeLa language to show its capabilities. These are the first efforts toward multidisciplinary monitoring of the oceans, which can extend beyond acoustic applications. [ABSTRACT FROM AUTHOR]

This article focuses on forthcoming research papers to be published in this periodical. One paper "On the Application of Factor Graphs and the Sum-Product Algorithm to ISI Channels," is based on the application of the sum-product algorithm to factor graphs representing the joint a posteriori probability of the transmitted symbols, we propose new iterative soft-input soft-output detection schemes for intersymbol interference channels. Another research called "A Class of Signal Processing Schemes for Reducing the Envelope Fluctuations of CDMA Signals," present a class of nonlinear signal processing schemes to reduce the envelope fluctuations in the downlink transmission within code division multiple access (CDMA) systems. These schemes are compatible with conventional CDMA receivers since they just require some modifications at the transmitter side. The proposed transmitter structure includes a nonlinear operation in the time-domain followed by a linear, frequency-domain filtering operation.