A New Adaptive OMP-MAP Algorithm-based Iterative Sparse Channel Estimation for OFDM Underwater Communication

A sparse channel estimation approach based on doubly spread underwater acoustic (UWA) channels is widely used todetect coherent acoustic orthogonal frequency division multiplexing (OFDM) signals. A new time-domain channelestimation (CE) technique for OFDM based UWA communication with Rician fading is used to exploit the channel sparsity.First, to improve the estimation accuracy in high noise conditions, we have exploited the channel sparsity to generate aclosed-form equation for the termination condition. Then, in low-level noise instances, the additional criterion to balanceestimation accuracy and computing costs has been established. By incorporating these two requirements within theorthogonal-matching-pursuit (OMP) structure, an adaptive-OMP (AOMP) algorithm has been proposed. The AOMP andmaximum a posteriori probability (MAP) techniques are combined to provide a computationally efficient, and a newAOMP-MAP scheme for estimating the sparse complex channel path gain has been proposed. Further, The minimumvariance unbiased estimator is used to improve the proposed CE technique. Exploiting the experimental channel data,computer simulations reveal that the proposed CE technique obtains the outstanding outcomes

Sparse channel estimation for multicarrier underwater acoustic communication : from subspace methods to compressed sensing

Author Posting. © IEEE, 2009. This article is posted here by permission of IEEE for personal use, not for redistribution. The definitive version was published in IEEE Transactions on Signal Processing 58 (2010): 1708-1721, doi:10.1109/TSP.2009.2038424.In this paper, we investigate various channel estimators that exploit channel sparsity in the time and/or Doppler domain for a multicarrier underwater acoustic system. We use a path-based channel model, where the channel is described by a limited number of paths, each characterized by a delay, Doppler scale, and attenuation factor, and derive the exact inter-carrierinterference (ICI) pattern. For channels that have limited Doppler spread we show that subspace algorithms from the array processing literature, namely Root-MUSIC and ESPRIT, can be applied for channel estimation. For channels with Doppler spread, we adopt a compressed sensing approach, in form of Orthogonal Matching Pursuit (OMP) and Basis Pursuit (BP) algorithms, and utilize overcomplete dictionaries with an increased path delay resolution. Numerical simulation and experimental data of an OFDM block-by-block receiver are used to evaluate the proposed algorithms in comparison to the conventional least-squares (LS) channel estimator.We observe that subspace methods can tolerate small to moderate Doppler effects, and outperform the LS approach when the channel is indeed sparse. On the other hand, compressed sensing algorithms uniformly outperform the LS and subspace methods. Coupled with a channel equalizer mitigating ICI, the compressed sensing algorithms can effectively handle channels with significant Doppler spread.C. Berger, S. Zhou, and P. Willett are supported by ONR grants N00014-09-10613, N00014-07-1-0805, and N00014-09-1-0704

MMP-DCD-CV based Sparse Channel Estimation Algorithm for Underwater Acoustic Transform Domain Communication System

In this paper, we propose a computationally efficient multipath matching pursuit (MMP) channel estimation algorithm for underwater acoustic (UWA) transform domain communication systems (TDCSs). The algorithm, referred to as the MMP-DCD-CV algorithm, is based on the dichotomous coordinate descent (DCD) iterations and cross validation (CV). The MMP-DCD-CV sparse channel estimator in each iteration searches for multiple promising path candidates most relevant to a residual vector and chooses the best candidate. The DCD iterations are used to solve the corresponding least squares problem with low complexity and numerical stability. The CV provides a stopping criterion of the algorithm without a priori information on the channel sparsity and noise level and examines whether the algorithm overfits its data, thus improving the estimation accuracy. The performance of the proposed algorithm is evaluated under simulated sparse UWA channels. The numerical results show that the algorithm achieves better performance than the original MMP algorithm, has lower complexity, and does not require prior knowledge on the channel sparsity and noise level. We also propose an UWA TDCS with sparse channel estimation based on the proposed MMP-DCD-CV algorithm. The proposed UWA communication system is tested by the Waymark simulator, providing the virtual signal transmission in the UWA channel, with a measured Sound Speed Profile and bathymetry. Numerical results demonstrate that the UWA TDCS with the proposed sparse channel estimator offers considerable improvement in system performance compared to other TDCS schemes

Adaptive OFDM Modulation for Underwater Acoustic Communications: Design Considerations and Experimental Results

Cataloged from PDF version of article.In this paper, we explore design aspects of adaptive modulation based on orthogonal frequency-division multiplexing (OFDM) for underwater acoustic (UWA) communications, and study its performance using real-time at-sea experiments. Our design criterion is to maximize the system throughput under a target average bit error rate (BER). We consider two different schemes based on the level of adaptivity: in the first scheme, only the modulation levels are adjusted while the power is allocated uniformly across the subcarriers, whereas in the second scheme, both the modulation levels and the power are adjusted adaptively. For both schemes we linearly predict the channel one travel time ahead so as to improve the performance in the presence of a long propagation delay. The system design assumes a feedback link from the receiver that is exploited in two forms: one that conveys the modulation alphabet and quantized power levels to be used for each subcarrier, and the other that conveys a quantized estimate of the sparse channel impulse response. The second approach is shown to be advantageous, as it requires significantly fewer feedback bits for the same system throughput. The effectiveness of the proposed adaptive schemes is demonstrated using computer simulations, real channel measurements recorded in shallow water off the western coast of Kauai, HI, USA, in June 2008, and real-time at-sea experiments conducted at the same location in July 2011. We note that this is the first paper that presents adaptive modulation results for UWA links with real-time at-sea experiments. © 2013 IEEE

Investigation of mode filtering as a preprocessing method for shallow-water acoustic communications

Author Posting. © IEEE, 2010. This article is posted here by permission of IEEE for personal use, not for redistribution. The definitive version was published in IEEE Journal of Oceanic Engineering 35 (2010): 744-755, doi:10.1109/JOE.2010.2045444.Acoustical array data from the 2006 Shallow Water Experiment (SW06) was analyzed to show the feasibility of broadband mode decomposition as a preprocessing method to reduce the effective channel delay spread and concentrate received signal energy in a small number of independent channels. The data were collected by a vertical array, which spans the water column from 12-m depth to the bottom in shallow water 80 m in depth. Binary-sequence data were used to phase-shift-keyed (PSK) modulate signals with different carrier frequencies. No error correction coding was used. The received signals were processed by a system that does not use training or pilot signals. Signals received both during periods of ordinary internal wave activity and during a period with unusually strong internal wave solitons were processed and analyzed. Different broadband mode-filtering methods were analyzed and tested. Broadband mode filtering decomposed the received signal into a number of independent signals with a reduced delay spread. The analysis of signals from the output of mode filters shows that even a simple demodulator can achieve a low bit error rate (BER) at a distance 19.2 km.This work was supported by the U.S. Office of Naval Research (ONR)

Design and implementation of low complexity wake-up receiver for underwater acoustic sensor networks

This thesis designs a low-complexity dual Pseudorandom Noise (PN) scheme for identity (ID) detection and coarse frame synchronization. The two PN sequences for a node are identical and are separated by a specified length of gap which serves as the ID of different sensor nodes. The dual PN sequences are short in length but are capable of combating severe underwater acoustic (UWA) multipath fading channels that exhibit time varying impulse responses up to 100 taps. The receiver ID detection is implemented on a microcontroller MSP430F5529 by calculating the correlation between the two segments of the PN sequence with the specified separation gap. When the gap length is matched, the correlator outputs a peak which triggers the wake-up enable. The time index of the correlator peak is used as the coarse synchronization of the data frame. The correlator is implemented by an iterative algorithm that uses only one multiplication and two additions for each sample input regardless of the length of the PN sequence, thus achieving low computational complexity. The real-time processing requirement is also met via direct memory access (DMA) and two circular buffers to accelerate data transfer between the peripherals and the memory. The proposed dual PN detection scheme has been successfully tested by simulated fading channels and real-world measured channels. The results show that, in long multipath channels with more than 60 taps, the proposed scheme achieves high detection rate and low false alarm rate using maximal-length sequences as short as 31 bits to 127 bits, therefore it is suitable as a low-power wake-up receiver. The future research will integrate the wake-up receiver with Digital Signal Processors (DSP) for payload detection. --Abstract, page iv

Efficient use of space-time clustering for underwater acoustic communications

Underwater acoustic (UWA) communication channels are characterized by the spreading of received signals in space (direction of arrival) and in time (delay). The spread is often limited to a small number of space-time clusters. In this paper, the spacetime clustering is exploited in a proposed receiver designed for guard-free orthogonal frequency-division multiplexing (OFDM) with superimposed data and pilot signals. For separation of space clusters, the receiver utilizes a vertical linear array (VLA) of hydrophones, whereas for combining delay-spread signals within a space cluster, a time-domain equalizer is used. We compare a number of space-time processing techniques, including a proposed reduced-complexity spatial filter, and show that techniques exploiting the space-time clustering demonstrate an improved detection performance. The comparison is done using signals transmitted by a moving transducer, and recorded on a 14-element non-uniform VLA in sea trials at distances of 46 km and 105 km

Performance evaluation of T-transform based OFDM in underwater acoustic channels

PhD ThesisRecently there has been an increasing trend towards the implementation of orthogonal frequency division multiplexing (OFDM) based multicarrier communication systems in underwater acoustic communications. By dividing the available bandwidth into multiple sub-bands, OFDM systems enable reliable transmission over long range dispersive channels. However OFDM is prone to impairments such as severe frequency selective fading channels, motioned induced Doppler shift and high peak-to-average-power ratio (PAPR). In order to fully exploit the potential of OFDM in UWA channels, those issues have received a great deal of attention in recent research. With the aim of improving OFDM's performance in UWA channels, a T-transformed based OFDM system is introduced using a low computational complexity T-transform that combines the Walsh-Hadamard transform (WHT) and the discrete Fourier transform (DFT) into a single fast orthonormal unitary transform. Through real-world experiment, performance comparison between the proposed T-OFDM system and conventional OFDM system revealed that T-OFDM performs better than OFDM with high code rate in frequency selective fading channels. Furthermore, investigation of different equalizer techniques have shown that the limitation of ZF equalizers affect the T-OFDM more (one bad equalizer coefficient affects all symbols) and so developed a modified ZF equalizer with outlier detection which provides major performance gain without excessive computation load. Lastly, investigation of PAPR reduction methods delineated that T-OFDM has inherently lower PAPR and it is also far more tolerant of distortions introduced by the simple clipping method. As a result, lower PAPR can be achieved with minimal overhead and so outperforming OFDM for a given power limit at the transmitter