In this paper, a frequency domain adaptive antenna array (FDAAA) algorithm is proposed for broadband single-carrier uplink transmissions in a cellular system. By employing AAA weight control in the frequency domain, the FDAAA receiver is able to suppress the multi-user interference (MUI) and the co-channel interference (CCI). In addition, the channel frequency selectivity can be exploited to suppress the inter-symbol interference (ISI) and to obtain frequency diversity (or the multi-path diversity). Another advantage of the FDAAA algorithm is that its performance is not affected by the spread of angles of arrival (AOA) of the received multi-path signal. In this study the structure of FDAAA receiver is discussed and the frequency domain signal-to-interference-plus-noise-ratio (SINR) after weight control is investigated. The performance of the FDAAA algorithm is confirmed by simulation results. It is shown that, the optimal FDAAA weight to obtain the best BER performance is that which fully cancels the interference when single-cell system is considered; On the other hand, when multi-cell cellular system is considered, the optimal FDAAA weight depends on both the cellular structure and the target signal to noise ratio (SNR) of transmit power control (TPC).

In this letter we present efficient iterative frequency domain equalization for single-carrier (SC) transmission systems with insufficient cyclic prefix (CP). Based on minimum mean square error (MMSE) criteria, iterative decision feedback frequency domain equalization (IDF-FDE) combined with cyclic prefix reconstruction (CPR) is derived to mitigate inter-symbol interference (ISI) and inter-carrier interference (ICI). Computer simulation results reveal that the proposed scheme significantly improves the performance of SC systems with insufficient CP compared with previous schemes.

The separation and localization of sound source signals are important techniques for many applications, such as highly realistic communication and speech recognition systems. These systems are expected to work without such prior information as the number of sound sources and the environmental conditions. In this paper, we developed a dodecahedral microphone array and proposed a novel separation method with our developed device. This method refers to human sound localization cues and uses acoustical characteristics obtained by the shape of the dodecahedral microphone array. Moreover, this method includes an estimation method of the number of sound sources that can operate without prior information. The sound source separation performances were evaluated under simulated and actual reverberant conditions, and the results were compared with the conventional method. The experimental results showed that our separation performance outperformed the conventional method.

In this paper, we propose a new differential MIMO single-carrier system with frequency-domain equalization (SC-FDE) aided by the insertion of cyclic prefix. This block transmission system not only inherits all the merits of the SISO SC-FDE system, but is also equipped with a differential space-time block coding (DSTBC) such as to combat the fast-changing frequency selective fading channels without the needs to estimate and then compensate the channel effects. Hence, for practical applications, it has the additional merits of decoding simplicity and robustness against high mobility transmission environments. Computer simulations show that the proposed system can provide diversity benefit as the non-differential system does, while greatly reducing the receiver complexity.

Error-propagation is an important issue and should be carefully coped with in the decision-feedback equalizers (DFE). Ignoring the impact of error-propagation often leads to impractical laboratory results. In this paper, we investigate two novel layered space-frequency equalizers (LSFE) for single-carrier multiple-input multiple-output (MIMO) systems, where the recently proposed frequency-domain equalizer with time domain noise-predictor (FDE-NP) is adopted at each stage of the LSFE. We first derive the partially-connected LSFE with noise predictor (PC-LSFE-NP) which has exactly the same mean square error (MSE) as the conventional LSFE under the assumption of perfect feedback. However, if error-propagation is considered, the proposed PC-LSFE-NP can achieve better performance than the conventional LSFE due to the more reliable feedback output by the decoders. To reduce the interference from the not yet detected layers in the feedback section, we then introduce the fully-connected LSFE with noise predictor (FC-LSFE-NP), in which all layers are implicitly equalized within each stage and their decisions fed back internally. The powerful feedback filter of FC-LSFE-NP brings significant performance superiority over the conventional LSFE and PC-LSFE-NP with either perfect or imperfect feedback. Moreover, we propose a simple soft-demapper for the equalizers to avoid information loss during decoding, and thus, further improve the performance. Finally, we compare the performance of (PC/FC)-LSFE-NP with the existing schemes by computer simulations.

This letter proposes a windowing frequency domain adaptive algorithm, which reuses the filtering error to apply window function in the filter updating symmetrically. By using a proper window function to reduce the negative influence of the spectral leakage, the proposed algorithm can significantly improve the performance of the acoustic echo cancellation for speech signals.

This paper proposes a frequency domain nulling filter and Turbo equalizer to suppress interference in the uplink of one-cell reuse single-carrier time division multiple access (TDMA) systems. In the proposed system, the desired signal in a reference cell is interfered by interference signals including adjacent-channel interference (ACI), co-channel interference (CCI), and intersymbol interference (ISI). At the transmitter, after a certain amount of spectrum is nulled considering the expected CCI, the suppressed power due to nulling is reallocated to the remaining spectrum components so as to keep the total transmit power constant. In this process, when mitigation of ACI is necessary, after a certain amount of spectrum at both edges is nulled using an edge-removal filter, the aforementioned process is conducted. At the receiver, frequency domain SC/MMSE Turbo equalizer (FDTE) is employed to suppress ISI due to spectrum nulling process in the transmitter as well as the multipath fading. Computer simulations confirm that the proposed scheme is effective in suppression of CCI, ACI and ISI in one-cell reuse single-carrier TDMA systems.

This paper proposes a novel channel estimation method for iterative equalization in MIMO systems. The proposed method incorporates co-channel interference (CCI) cancellation in the channel estimator and the channel estimation is successively performed with respect to each stream. Accuracy of channel estimation holds the key to be successfully converged the iterative equalization and decoding process. Although the channel estimates can be re-estimated by means of LS (Least Square) channel estimation using tentative decisions obtained in the iterative process, its performance is severely limited in a MIMO system because of erroneous decisions and ill-conditioned channel estimation matrix. The proposed method can suppress the above effects by means of CCI cancellation and successive channel estimation. Computer simulation confirms that the proposed channel estimation method can accurately estimate the channel, and the receiver with iterative equalization and the proposed method achieves excellent decoding performance in a MIMO-SM system.

The frequency domain binaural model (FDBM) has been previously proposed to localize multiple sound sources. Since the method requires only two input signals and uses interaural phase and level differences caused by the diffraction generated by the head, flexibility in application is very high when the head is considered as an object. When an object is symmetric with respect to the two microphones, the performance of sound source localization is degraded, as a human being has front-back confusion due to the symmetry in a median plane. This paper proposes to reduce the degradation of performance on sound source localization by a combination of the microphone pair outputs using the FDBM. The proposed method is evaluated by applying to a security camera system, and the results showed performance improvement in sound source localization because of reducing the number of cones of confusion.

A binaural hearing assistance system based on the frequency domain binaural model has been previously proposed. The system can enhance a signal coming from a specific direction. Since the system utilizes a binaural signal, an inter-channel communication between left and right subsystems is required. The bit rate reduction in inter-channel communication is essential for the detachment of the headset from the processing system. In this paper, the performance of a system which uses a differential pulse code modulation codec is examined and the relationship between the bit rate and sound quality is discussed.

We design a unified multicarrier (UMC) system for wideband communication. The proposed scheme can provide an effective and unified method that can implement a wideband CDMA system with high spectrum efficiency and flexibility because of the free selection of system parameters and a double spreading in the time and frequency domains. Also, separation of the spectrums carrying the same data to further ensure the independent fading between subcarriers is performed, that is, subcarriers are interleaved in the frequency domain. This frequency interleaving mitigates the effect of ISI and ICI. We also theoretically analyze the performance of the UMC system by deriving the closed-form solution for probability of bit error in a frequency selective Rayleigh fading channel. The analysis has proved that the UMC system has outperformed the conventional single carrier CDMA system under given conditions.

A multiple-antenna receiving and combining scheme is proposed for high-data-rate transmitted-reference (TR) Ultra-Wideband (UWB) systems. The nonlinearity of the inter-symbol interference (ISI) model is alleviated via simple antenna combining. Under the simplified ISI model, frequency domain equalization (FDE) is adopted and greatly reduces the complexity of the equalizer. A simple estimation algorithm for the simplified ISI model is presented. Simulation results demonstrate that compared to the single receive antenna scheme, the proposed method can obtain a significant diversity gain and eliminate the BER floor effect. Moreover, compared to the complex second-order time domain equalizer, FDE showed better performance robustness in the case of imperfect model estimation.

MIMO (Multiple-Input Multiple-Output) technologies have attracted much interest for high-rate and high-capacity wireless communications. MIMO technologies under frequency-selective fading environments (wideband MIMO technologies) have also been studied. A wideband MIMO system is affected by ISI (Inter Symbol Interference) and CCI (Co-Channel Interference). Hence, we need a MIMO signal detection technique that simultaneously suppresses ISI and CCI. The OFDM system and SC-FDE (Single Carrier-Frequency Domain Equalization) techniques are often used for suppressing ISI. By employing these techniques with the ZF (Zero Forcing) or the MMSE (Minimum Mean Square Error) spatial filtering technique, we can cancel both ISI and CCI. To use ZF or MMSE, we need channel state information for calculating the receive weights. Although an LS (Least Square) channel estimation technique has been proposed for MIMO-OFDM systems, it needs a large estimation matrix at the receiver side to obtain sufficient estimation performance in heavy multipath environments. However, the use of a large matrix increases computational complexity and the circuit size. We use frequency domain channel estimation to solve these problems and propose an iterative method for achieving better estimation performance. In this paper, we assume the use of a MIMO-UWB system that employs a UWB-IR (Ultra-Wideband Impulse Radio) scheme with the FDE technique as the wideband wireless transmission scheme for heavy multipath environments, and we evaluate the iterative frequency domain channel estimation through computer simulations and computational complexity calculations.

This paper proposes low-complexity pre- and post-frequency domain equalization and frequency diversity combining methods for block transmission schemes with cyclic prefix. In the proposed methods, the equalization and diversity combining are performed simultaneously in discrete frequency domain. The weights for the proposed equalizer and combiner are derived based on zero-forcing and minimum-mean-square error criteria. We demonstrate the performance of the proposed methods, including bit-error rate performance and peak-to-average power ratios of the transmitted signal, via computer simulations.

This paper describes a method of free iris and focus image generation based on transformation integrating multiple differently focused images. First, we assume that objects are defocused by a geometrical blurring model. And we combine acquired images on certain imaging planes and spatial information of objects by using a convolution of a three-dimensional blur. Then, based on spatial frequency analysis of the blur, we design three-dimensional filters that generate free iris and focus images from the acquired images. The method enables us to generate not only an all-in-focus image corresponding to an ideal pin-hole iris but also various images, which would be acquired with virtual irises whose sizes are different from the original one. In order to generate a certain image by using multiple differently focused images, especially very many images, conventional methods usually analyze focused regions of each acquired image independently and construct a depth map. Then, based on the map, the regions are merged into a desired image with some effects. However, generally, it is so difficult to conduct such depth estimation robustly in all regions that these methods cannot prevent merged results from including visible artifacts, which decrease the quality of generated images awfully. In this paper, we propose a method of generating desired images directly and robustly from very many differently focused images without depth estimation. Simulations of image generation are performed utilizing synthetic images to study how certain parameters of the blur and the filter affect the quality of generated images. We also introduce pre-processing that corrects the size of acquired images and a simple method for estimating the parameter of the three-dimensional blur. Finally, we show experimental results of free iris and focus image generation from real images.

In order to improve the forward link capacity of cdma2000 HRPD (High Rate Packet Data) or CDMA2000 1xEV-DO, it is significant to overcome multi-path interference. This paper focuses on FDE (Frequency Domain Equalization) with MMSE (Minimum Mean Square Error) criterion. On top of that, backward compatibility with HRPD should be maintained, in other words common channels such as the pilot channel should not be changed. Thus, the PN (Pseudo Noise) spread pilot block without CP (Cyclic Prefix) signals has to be dealt with for FDE. However, this will cause the conventional channel estimation accuracy to deteriorate. In order to improve the estimation accuracy of the conventional method, this paper presents a MRC (Maximal Ratio Combining) spectrum estimator, IPI (Inter-Path Interference) canceller, and path searcher. The results obtained from computer simulations reveal that the proposed method can improve the PER (Packet Error Rate) performance significantly. If compared with Rake combiner and TDE (Time Domain Equalization) with NLMS (Normalized Least Mean Square) scheme, the maximum data rates at a fixed PER of 1% can be increased by 5 to 8 times and 1.25 to 2.67 times, respectively.

In this letter two noise-predictive equalization schemes, which are based on zero-forcing (ZF) and minimum mean square error (MMSE) criteria, are presented for unique-word (UW) based single-carrier systems. The correlation properties of the noises in the outputs of the frequency domain equalizer are exploited to predict and cancel the noise contained in the estimation of data. Theoretical analyses show that both of the proposed techniques perform better than the conventional frequency domain equalizers. Simulation results have confirmed the significant performance improvement they could achieve.

A novel frequency domain training sequence and the corresponding carrier frequency offset (CFO) estimator are proposed for orthogonal frequency division multiplexing (OFDM) systems over frequency-selective fading channels. The proposed frequency domain training sequence comprises two types of pilot tones, namely distinctively spaced pilot tones with high energies and uniformly spaced ones with low energies. Based on the distinctively spaced pilot tones, integer CFO estimation is accomplished. After the subcarriers occupied by the distinctively spaced pilot tones and their adjacent subcarriers are nulled for the sake of interference cancellation, fractional CFO estimation is executed according to the uniformly spaced pilot tones. By exploiting a predefined lookup table making the best of the structure of the distinctively spaced pilot tones, computational complexity of the proposed CFO estimator can be decreased considerably. With the aid of the uniformly spaced pilot tones generated from Chu sequence with cyclically orthogonal property, the ability of the proposed estimator to combat multipath effect is enhanced to a great extent. Simulation results illustrate the good performance of the proposed CFO estimator.

The speech noise reduction system based on the frequency domain adaptive line enhancer using a windowed modified DFT (MDFT) pair is presented. The adaptive line enhancer (ALE) is effective for extracting sinusoidal signals blurred by a broadband noise. In addition, it utilizes only one microphone. Therefore, it is suitable for the realization of speech noise reduction in portable electronic devices. In the ALE, an input signal is generated by delaying a desired signal using the decorrelation parameter, which makes the noise in the input signal decorrelated with that in the desired one. In the present paper, we propose to set decorrelation parameters in the frequency domain and adjust them to optimal values according to the relationship between speech and noise. Such frequency domain decorrelation parameters enable the reduction of the computational complexity of the proposed system. Also, we introduce the window function into MDFT for suppressing spectral leakage. The performance of the proposed noise reduction system is examined through computer simulations.

This paper presents a study on the blind separation of a convoluted mixture of speech signals using Frequency Domain Independent Component Analysis (FDICA) algorithm based on the negentropy maximization of Time Frequency Series of Speech (TFSS). The comparative studies on the negentropy approximation of TFSS using generalized Higher Order Statistics (HOS) of different nonquadratic, nonlinear functions are presented. A new nonlinear function based on the statistical modeling of TFSS by exponential power functions has also been proposed. The estimation of standard error and bias, obtained using the sequential delete-one jackknifing method, in the approximation of negentropy of TFSS by different nonlinear functions along with their signal separation performance indicate the superlative power of the exponential-power-based nonlinear function. The proposed nonlinear function has been found to speed-up convergence with slight improvement in the separation quality under reverberant conditions.