Impulse Response Shortening Research Articles

Exponential excitations for effective identification of Wiener system

The paper considers the problem of nonparametric estimation of nonlinear characteristic in the FIR Wiener system. Methods proposed so far have suffered from the so-called ‘curse of dimensionality’ and their practical applications have been limited to the very short impulse responses of the linear component. In the proposed approach, the class of (stochastically initialised) repeated exponential input excitations is introduced, and a modified kernel-based estimator is presented. It is shown that the rate of convergence to the genuine system characteristic is significantly improved and does not depend on the length of the impulse response of the linear dynamic block.

On Inclusive Growth: An Economic Growth Time-Frequency Analysis of the Democratic Republic of Congo from 1975-2016

This paper focuses on fluctuations in the inclusiveness of economic growth (in terms of its participatory aspect) in the Democratic Republic of Congo at different time scales. We investigate the extent to which variations in economic growth are explained by variations in the unemployment rate, the agricultural sector, and the mining sector, using an econometric model known as Auto-Regressive Distributed Lag model which will give us both short and long-term impulse responses in order to identify the dynamics of our variables (endogenous and exogenous). In the same vein, we show that the development of a structural economic policy is imperative to overcome the problem of the non-resilience of the Congolese economy, in particular through the promotion of growth-catalyzing sectors in order to generate more wealth to be equitably redistributed to each segment of the Congolese population as well as to the pro-poor sectors. Moreover, knowing that the Congolese economy has been confronted with multiple shocks (supply and demand) to the point of damaging its sustainability, the information contained in the fluctuations of economic growth informs us about the state of inclusive growth in its entirety relative to the interference of the Congolese state. Thus, the cyclical movements of economic growth are extracted using a multi-resolution analysis (or wavelet decomposition) that spreads a time series at different levels of resolution.

Impulse Response Shortening in TDOA Algorithm for Underwater Acoustic Localization

This paper presents a new idea of impulse response shortening in time difference of arrival (TDOA) measurement algorithm for underwater acoustic localization. Due to strong multipath interference caused by acoustic wave reflection, the pseudo-peaks occur when calculating a correlation function. Their pseudo-peaks disturbs the detection of a correct time position and leads to large TDOA measurement errors. In our previous paper, we presented the impulse response based GCC-PHAT (IR-GCC-PHAT), which computes a time difference by taking a cross-correlation between two impulse responses. As the further improvement, we study the effects of impulse response shortening that removes the pseudo-peaks. The effectiveness of impulse response shortening is explained by observing acoustic wave paths on an acoustic propagation simulator and verified by the simulation results

AN EMPIRICAL EVIDENCE ON THE IMPACT OF REAL EFFECTIVE EXCHANGE RATE ON TRADE BALANCE

This study aimed to examine whether the real effective exchange rate has an impact on trade balance in short run and long run, as well as to investigate if the Marshall-Lerner condition and J-curve phenomenon is satisfied by the variables in Malaysia from 2010 until 2019. Monthly data was used where the dependent variable was trade balance and the independent variable was real effective exchange rate. Several econometric models are applied including the Johansen co-integration to investigate long run relationship and Marshall-Lerner condition estimation, VECM model to investigate short run relationship estimation and impulse response test for J-curve phenomenon estimation. The findings revealed that there was long run relationship between the variables and there was existence of Marshall-Lerner condition. Meanwhile, there was no short run relationship between the variables and J-curve phenomenon was not satisfied in this study. This could be due to devaluation of real effective exchange rate where will help to improve the trade balance in the long run but give a negative effect in the short run which not followed the J-curve pattern. Hence, the Malaysia government should take high concern of exchange rate policy as well as monetary and fiscal policy in order to help stimulate the exports and reduce the inflationary pressure due to devaluation of exchange rate.

OPTIMIZATION-BASED RECURSIVE FILTERING FOR VIBROSEIS HARMONIC NOISE ELIMINATION

The vibroseis sources always produce harmonics, which are considered here as noise to be eliminated. An algorithm for removing such noise has been developed based on the previously obtained mathematical model of a vibroseis wavelet distorted by the harmonics. It is shown that the ideal inversion operator is recursive, i.e. is a filter with infinite impulse response. However, due to the peculiarity of the problem, this operator is successfully approximated by a filter with a short impulse response, allowing linearization of the optimization scheme. An objective being the energy of the noise attenuation result is formed. It allows application of the geometrical divergence correction despite the fact that it introduces distortions in the signal shape and the harmonics.

Violin mpulse esponse ength and erceptions of cceptability

The design, execution and analysis of a double-blind listening study is described, in which participants gave preference ratings for nine versions of the same piece of music, obtained by convolving violin impulse responses, of different length, with the original piece played on an electric instrument and stored in digital form. The original impulse response, with a length of 91.4 ms was measured from a Stradivarius violin and progressively degraded by shortening its length. The participants, who were all trained musicians, were asked to record their preferences based on personal taste, not perceived measures of quality. Analysis of the data revealed a sigmoid relationship between preference and length of impulse response. Emulated music generated using short impulse responses was the least preferred, and this aversion was reasonably constant for responses shorter than 2.7 ms. However, and perhaps surprisingly, impulse responses of only 5.78 ms were deemed acceptable. Extending the length beyond this value had little effect on the attributed preferences. Between these two values, there was a steep increase in the assigned scores, suggesting a sigmoid relationship. The findings are of theoretical interest for psychoacoustics and can be applied to the development of electronic devices that emulate stringed instruments in real time.

Design of Modulated Filter Banks and Transmultiplexers With Unified Initial Solutions and Very Few Unknown Parameters

This brief proposes a method for designing modulated filter banks (FBs) with a large number of channels. The impulse response of the long prototype filter is parameterized in terms of a few short impulse responses, thus significantly reducing the number of unknown parameters. The proposed method starts by first obtaining an FB with a few channels. The solution of this FB is then partly reused as an initial (very close to final) solution in the design of FBs with a large number of channels. The number of unknown parameters hence drastically reduces. For example, we can first design a cosine modulated FB (CMFB) with three channels whose prototype filter has a stopband attenuation of about 40 dB. We can then reuse the solution of this CMFB in the design of a CMFB with 16 384 channels whose prototype filter has a similar stopband attenuation. With our proposed method, we need to reoptimize only 14 parameters to design the CMFB with 16 384 channels.

Automatic evaluation of the fundamental frequency variations and related damping factor of reinforced concrete framed structures using the Short Time Impulse Response Function (STIRF)

Structural Health Monitoring (SHM) aims to improve knowledge of the safety and maintainability of civil structures and infrastructures. This paper presents an innovative strategy for automatic evaluation of the variable fundamental frequency and related damping factors of nonlinear structures during strong motion phases. In fact, most of methods for damage detection are based on the assessment of the variations related to dynamic parameters characterizing the monitored structure. In this case, a crucial aspect is a correct estimation of both structural eigenfrequencies and related damping factor also during the nonlinear behavior of a monitored structure. In this paper, nonlinear interferometric analyses combined with the Fourier Transform (FT) are proposed to evaluate the Short-Time Impulse Response Function (STIRF) able to characterize frequencies and damping variations of a monitored structure. Two application of the STIRF are proposed on both numerical and experimental models.

DMT Transmission in SI POF With Minimax Channel Shortening Equalizer

Discrete multitone (DMT) is a viable candidate for modulation format in step index polymere optical fibers (SI POFs). In this letter, it is shown that the performance of the DMT system in SI POF can be considerably improved by using a time-domain impulse response shortening equalizer. A new method of calculating the equalizer's coefficients, which employs minimax optimization, is formulated. The proposed method gives similar or even slightly better results than the eigen filter method used as reference. Increase of bit rate by 7.1 (9.5)% in 50 (100) m SI POF is experimentally confirmed using the minimax method.

Shaping Quantization Noise and Clipping Distortion in Direct-Detection Discrete Multitone

This paper introduces a method for boosting the signal-to-noise ratio at the transmitter of a discrete-multitone system through reshaping of the spectrum of quantization noise and clipping distortion. This is of particular importance for high-bandwidth transmission, such as 30 GHz, where digital-to-analog converters have a very limited resolution of only 5-8 bits. This limitation of the data converters is thus significant and reducing their noise contribution has a pronounced effect on the overall system's performance. The proposed technique resembles the quantization noise shaping of delta-sigma data converters but is a more general technique that shapes both quantization and clipping noise and it additionally does not require oversampling. The technique relies on the existence of an unused band of subcarriers, as would be the case, for example, when tone reservation is applied. The proposed system is additionally well-suited for low-power transceivers as its most computationally-expensive element is a short finite impulse response filter. In a practical setting, the noise contribution of quantization and clipping can be reduced by 6 dB.

Generalization of Multi-Channel Linear Prediction Methods for Blind MIMO Impulse Response Shortening

The performance of many microphone array processing techniques deteriorates in the presence of reverberation. To provide a widely applicable solution to this longstanding problem, this paper generalizes existing dereverberation methods using subband-domain multi-channel linear prediction filters so that the resultant generalized algorithm can blindly shorten a multiple-input multiple-output (MIMO) room impulse response between a set of unknown number of sources and a microphone array. Unlike existing dereverberation methods, the presented algorithm is developed without assuming specific acoustic conditions, and provides a firm theoretical underpinning for the applicability of the subband-domain multi-channel linear prediction methods. The generalization is achieved by using a new cost function for estimating the prediction filter and an efficient optimization algorithm. The proposed generalized algorithm makes it easier to understand the common background underlying different dereverberation methods and future technical development. Indeed, this paper also derives two alternative dereverberation methods from the proposed algorithm, which are advantageous in terms of computational complexity. Experimental results are reported, showing that the proposed generalized algorithm effectively achieves blind MIMO impulse response shortening especially in a mid-to-high frequency range.

Design and Test of an Imaging System for UWB Breast Cancer Detection

Electromagnetic ultra-wideband (UWB) sensing and imaging provide perspectives for early-stage breast cancer detection. This paper deals with practical chal- lenges of real measurements. We present an experimental setup for breast phantom trials based on M-sequence radar technology and short active dipole antennas. It com- bines short impulse responses, appropriate fidelity and very small antenna dimension and allows array construc- tion with sufficient number of antennas around the breast. The basic approach and obtained imaging results are pre- sented. Furthermore, in this extended paper version con- tinuative development steps are described and measure- ment results reflecting specific performance aspects are discussed.

Generic Feasibility of Perfect Reconstruction With Short FIR Filters in Multichannel Systems

We study the feasibility of short finite impulse response (FIR) synthesis for perfect reconstruction (PR) in generic FIR filter banks. Among all PR synthesis banks, we focus on the one with the minimum filter length. For filter banks with oversampling factors of at least two, we provide prescriptions for the shortest filter length of the synthesis bank that would guarantee PR almost surely. The prescribed length is as short or shorter than the analysis filters and has an approximate inverse relationship with the oversampling factor. Our results are in form of necessary and sufficient statements that hold generically, hence only fail for elaborately-designed nongeneric examples. We provide extensive numerical verification of the theoretical results and demonstrate that the gap between the derived filter length prescriptions and the true minimum is small. The results have potential applications in synthesis FB design problems, where the analysis bank is given, and for analysis of fundamental limitations in blind signals reconstruction from data collected by unknown subsampled multi-channel systems.

Cyclic Prefix Design and Allocation in Bit-Loaded OFDM over Power Line Communication Channels

The usual approach when designing an orthogonal frequency division multiplexing (OFDM) system, is to dimension the length of the cyclic prefix (CP) equal to the length of a typical bad (i.e., long) channel impulse response, so that both inter-symbol and inter-carrier interference are avoided for almost all channel realizations. However, such an approach does not maximize system capacity. It (a) wastes channel resources for relatively short channel impulse response realizations and (b) it is not necessarily optimal to completely eliminate interference. In this paper, we study the problem of designing the CP length for OFDM systems in a capacity-optimal way. To this end, we first present optimal and simplified metrics suitable to maximize capacity. Then, we apply those metrics and propose practical resource (bit-loading and OFDM sub-channel assignment) algorithms that include CP-length adaptation. We present numerical results for the example of a power line communication (PLC) OFDM system with typical indoor PLC channels that confirm the gains achievable with the proposed CP-length adaptation.

Room Impulse Response Shortening/Reshaping With Infinity- and $p$-Norm Optimization

The purpose of room impulse response (RIR) shortening and reshaping is usually to improve the intelligibility of the received signal by prefiltering the source signal before it is played with a loudspeaker in a closed room. In an alternative, but mathematically equivalent setting, one may aim to postfilter a recorded microphone signal to remove audible echoes. While least-squares methods have mainly been used for the design of shortening/reshaping filters for RIRs until now, we propose to use the infinity- or p-norm as optimization criteria. In our method, design errors will be uniformly distributed over the entire temporal range of the shortened/reshaped global impulse response. In addition, the psychoacoustic property of masking effects is considered during the filter design, which makes it possible to significantly reduce the filter length, compared to standard approaches, without affecting the perceived performance.

Joint Channel and Echo Impulse Response Shortening for High-Speed Data Transmission

The unit-norm constraint optimization for joint shortening of channel and echo impulse response is presented in this paper. The optimization is performed in the mean-square sense (MSE) and compared to unit-tap constraint joint shortening optimization. The proposed method could serve in performing joint echo cancellation and equalization for data transmission over category cables for the next IEEE standard on short-range ultra high-speed copper interconnections used in data servers, high-performance computing centers, local area networks, etc. The main objective of the presented algorithm is to extensively reduce the power and implementation complexities of a 40GBASE-T system.

An Overlap-Save Frequency-Domain Implementation of the Delayless Subband ANC Algorithm

An efficient active noise control algorithm based on the delay less subband adaptive filter was previously proposed by Park (ldquoA delayless subband active noise control system for wideband noise control, <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">IEEE</i> <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">transactions</i> <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">on</i> <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">speech</i> <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">and</i> <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">audio</i> <i xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">processing</i> , vol. 9, no. 8, pp. 892-897, November 2001), whereby the computational complexity is reduced significantly by using a short impulse response filter to model the secondary path transfer function in a subband-decomposed form. In this correspondence, an overlap-save frequency-domain implementation of the algorithm is proposed. Simulations show that the proposed implementation has better performance than the original.

Reducing the Delay of Phasor Estimates Under Power System Oscillations

<para xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"> The delay and computational complexity of phasor estimates under power oscillations are minimized in this paper. Different approximations to the raised-cosine (RC) filter lead to a reduction in the delay and computational complexity of the phasor estimates without significant distortion. The minimum-phase implementation of the linear-phase RC filter reduces the delay of the phasor estimates by about half, with a shorter impulse response and without the introduction of significant phase distortion. This improvement in speed provides faster phasor estimates for real-time monitoring and control applications in power systems. On the other hand, infinite-impulse-response filters reduce the computational complexity of the estimates by a factor of ten but without significant delay reduction and with a longer transient behavior and the introduction of a higher distortion. However, this solution could be useful for implementations in which the concern about computational complexity prevails over the concern about delay and distortion. </para>

Cost-Effective Joint Echo-NEXT Canceller Designs for 10GBase-T Ethernet Systems Based on a Shortened Impulse Response Filter (SIRF) Scheme

Echo canceller plays an important role in the full-duplex communication system. Conventional implementations of echo cancellers are often the adaptive transversal filter architectures due to the simplicity and robustness of stability and convergence. However, the conventional echo cancellers suffer from high cost problem especially when the response time of the echo is long. In this paper, a new cost-efficient architecture of echo cancellers, targeting on 10GBase-T Ethernet System, is presented. The proposed scheme inherits the concept of channel shortening which is widely employed in DSL systems. A shortened impulse response filter is implemented at the receiver to shorten the impulse response of the echo signal. Hence, the overall cost of echo cancellers can be reduced. We generalize the channel shortening architecture to a joint multi-channel shortening scheme. The joint multi-channel shortening architecture can be applied to multiple-input multiple-output wireline communication systems to further reduce both the cost of echo and near-end crosstalk (NEXT) cancellers. We apply the proposed scheme to 10GBase-T Ethernet system. The simulation results show that the proposed echo and NEXT cancellers can save up to 35% hardware cost compared to the conventional transversal implementations.

A study on combining acoustic echo cancelers with impulse response shortening

In hands-free video conferencing systems acoustic echo cancelers (AECs) have to face the problem of very high-order impulse responses (IRs), which have to be compensated. Time-domain algorithms for adaptation often suffer from slow convergence (as the NLMS algorithm, e.g.) or high computational complexity (e.g., the RLS). On the other hand frequency-domain algorithms introduce undesired delays [S. Haykin, Filter Theory, 2002]. For high-quality hands-free systems IR shortening concepts and IR shaping concepts developed for listening room compensation (LRC) [M.Kallinger and A. Mertins, in Proc. Asilomar, 2005] can be applied to increase speech intelligibility for the near-end speaker. The aim of this study is the synergetic combination of LRC concepts with acoustic echo cancellation. For this scenario two different forms of concatenating the subsystems are possible. Either the AEC filter follows the LRC or vice versa. In the first case the equalization filter reduces the length of the effective IR seen by the AEC filter. Thus, shorter AEC filters can be used which results in faster convergence. However, an estimation algorithm for the room IR is necessary for the LRC subsystem. In the second case the AEC delivers an estimate of the room IR which can be used as an input for the LRC filter. Experimental results confirm the superiority of the new combined approach.