Dissertations / Theses on the topic 'Channel shortening'

Ultra Wideband (UWB) communication systems occupy large bandwidths with very low power spectral densities. This feature makes UWB channels highly rich in multipaths. To exploit the temporal diversity, a UWB receiver usually incorporates Rake reception. Each multipath in the channel carries just a fraction of the signal energy. This phenomenon dictates a Rake receiver with a large number of fingers to achieve good energy capture and output signal to noise ratio (SNR). Eventually, the Rake structure becomes very complex from analysis and design perspectives and incurs higher manufacturing cost. The first contribution of this thesis is to propose channel shortening or time domain equalization as a technique to reduce the complexity of the UWB Rake receiver. It is analyzed that most of the existing channel shortening equalizer (CSE) designs are either system specific or optimize a parameter not critical or even available in UWB systems. The CSE designs which are more generic and use commonly critical cost functions may perform poorly due to particular UWB channel profiles and related statistical properties. Consequently, the main contribution of the thesis is to propose several CSE designs to address the specific needs of UWB systems. These CSE designs not only exploit some general but also some UWB specific features to perform the task more efficiently. The comparative analysis of the proposed CSEs, some existing designs and the conventional Rake structures leads towards the conclusion. It is finally shown that the use of CSE at the receiver front end greatly simplifies the Rake structure and the associated signal processing.

A new time domain blind adaptive channel shortening algorithm for Discrete Multi Tone (DMT)-based multicarrier systems is first proposed. It is computationally less expensive, and more robust to non- Gaussian impulsive noise environments than a recently reported Sum squared Autocorrelation Minimization (SAM) algorithm. A "left" initialization scheme is also suggested for Carrier Serving Area (CSA) loop Asymmetric Digital Subscriber Line (ADSL) channels. Simulation studies show that by a proper selection of the learning parameter i.e., the step size, the bit rates achieved by the SAM algorithm when operating in an environment contaminated by Additive White Gaussian Noise (AWGN) can be further improved. Next a novel time domain low complexity blind adaptive channel short ening algorithm called Single Lag Autocorrelation Minimization (SLAM) is introduced. The algorithm is totally blind in the sense that it does not require a prior knowledge about the length of the channel impulse response. The proposed novel stopping criterion freezes the adaptation of the SLAM algorithm when the maximum amount of Inter Symbol Interference (ISI) is cancelled. As such, the stopping criterion can also be used with SAM. An attractive alternate frequency domain equalization approach for multicarrier systems is Per Tone Equalization (PTEQ). This scheme en- ables true signal-tonoise ratio optimization to be implemented for each tone and it always achieves higher bit rates than Time domain Equalizer (TEQ) based channel shortening schemes but at the price of increased computational complexity and higher memory requirements. A low complexity (PTEQ) scheme is, therefore, finally proposed. The com plexity of the PTEQ can be traded off with the complexity of the timing synchronization within the system. In particular, it is shown that the use of more than one difference terms and hence a long equalizer in the PTEQ scheme is generally redundant. The PTEQ scheme assumes knowledge of the channel impulse response. In this case synchronization is trivial and it is possible to use only a length two PTEQ equalizer and attain essentially identical bit rate performance to a PTEQ equalizer with length matched to the cyclic prefix. This observation allows for a substantial reduction in computational complexity of the PTEQ scheme in both initialization and data transmission modes. For a reasonable range of values of synchronization error, <5, around the optimal value of 5 = 0, the performance of this length two equalizer is shown to remain relatively constant. For positive synchronization errors, however, the required PTEQ equalizer length is proportional to the synchronization error. A low complexity blind synchronization method is ultimately suggested which is based on the construction of the difference terms of the PTEQ scheme.

In wireline multi carrier systems a cyclic prefix is commonly used to facilitate simple channel equalization at the receiver. The selection of the length of the cyclic prefix is a trade-off between maximizing the length of the channel for which inter-symbol interference is eliminated and optimizing the transmission efficiency. When the length of the channel exceeds that of the cyclic prefix, adaptive channel shorteners can be used to force the effective channel length of the combined channel and channel shortener to satisfy the cyclic prefix constraint. The focus of this thesis is the design of new blind adaptive time-domain algorithms for channel shortening in wireline multicarrier systems, with good convergence properties and low computational complexity.

In wireline multicarrier systems a cyclic prefix is generally used to facilitate simple channel equalization at the receiver. The choice of the length of the cyclic prefix is a trade-off between maximizing the length of the channel for which inter-symbol interference is eliminated and optimizing the transmission efficiency. When the length of the channel is greater than the cyclic prefix, adaptive channel shorteners can be used to force the effective channel length of the combined channel and channel shortener to be within the cyclic prefix constraint. The focus of this thesis is the design of new blind adaptive time-domain channel shortening algorithms with good convergence properties and low computational complexity. An overview of the previous work in the field of supervised partial update adaptive filtering is given. The concept of property-restoral based blind channel shortening algorithms is then introduced together with the main techniques within this class of adaptive filters. Two new partial update blind (unsupervised) adaptive channel shortening algorithms are therefore introduced with robustness to impulsive noise commonly present in wireline multicarrier systems. Two further blind channel shortening algorithms are proposed in which the set of coefficients which is updated at each iteration of the algorithm is chosen deterministically. One of which, the partial up-date single lag autocorrelation maximization (PUSLAM) algorithm is particularly attractive due to its low computational complexity. The interaction between the receiver matched filter and the channel shortener is considered in the context of a multi-input single-output environment. To mitigate the possibility of ill-convergence with the PUSLAM algorithm an entirely new random PUSLAM (RPUSLAM) algorithm is proposed in which randomness is introduced both into the lag selection of the cost function underlying SLAM and the selection of the particular set of coefficients updated at each algorithm. This algorithm benefits from robust convergence properties whilst retaining relatively low computational complexity. All algorithms developed within the thesis are supported by evaluation on a set of eight carrier serving area test loop channels.

Le contexte de ces travaux de thèse est la transmission dite faster-than-Nyquist (FTN). Cette technique propose d’augmenter l’efficacité spectrale en augmentant lerythme de transmission au-delà de la bande occupée par le signal émis, indépendamment de laconstellation choisie. Il a été montré que le FTN offre des taux d’information supérieurs à ceuxdes systèmes de Nyquist. Toutefois, le non respect du critère de Nyquist entraîne l’apparitiond’interférence entre symboles et des techniques de réception appropriées doivent être utilisées.La technique de réception dite channel shortening consiste à filtrer la séquence reçue puis àcalculer des probabilités symbole a posteriori approximatives à l’aide de l’algorithme BCJRen considérant une réponse de canal modifiée, de longueur réduite. Dans la littérature, enprésence d’information a priori, les filtres du récepteur channel shortening sont optimiséssous critère de maximisation de l’information mutuelle généralisée (IMG) en utilisant desméthodes numériques. Nous proposons dans ces travaux de thèse une solution analytiquepour l’ensemble des filtres channel shortening sous critère de maximisation de l’IMG lorsquele récepteur dispose d’information a priori. Nous démontrons ensuite que l’égaliseur au sens dela minimisation de l’erreur quadratique moyenne (MMSE) est un cas particulier de l’égaliseurchannel shortening. Dans le cadre de la turbo égalisation, nous étudions ensuite un estimateurpermettant d’obtenir l’information a priori à partir de l’information en sortie du décodeurcorrecteur d’erreurs. Finalement, nous évaluons les performances du système complet aveccodage correcteur d’erreurs sur canal à bruit additif blanc Gaussien
In order to increase the spectral efficiency of digital communications systems,the faster-than-Nyquist (FTN) approach increases the symbol rate beyond the occupied bandwidthof the transmitted signal independently of the constellation type and size. It has beenshown that information rates of FTN systems are greater than those of Nyquist systems.However, the non-compliance of the Nyquist criterion causes inter-symbol interference to appearand therefore appropriate reception techniques must be used. At reception, the channelshortening approach consists on a receiving filter followed by a BCJR algorithm computingapproximate a posteriori symbol probabilities by considering a modified channel response ofreduced length. In the literature, the channel shortening receiving filters are chosen to maximizethe generalized mutual information (GMI). Such optimization is performed by usingnumerical optimization methods. In this PhD thesis, we propose a closed-form solution forall channel shortening filters considering the GMI maximization criterion. We show that theminimum mean square error (MMSE) equalizer is a particular case of the channel shorteningapproach. Within the frame of turbo equalization, we then study a suitable estimator allowingto obtain symbols a priori information from the information provided by the a decoder. Finally,we study the performance of the complete system with channel coding over an additivewhite Gaussian noise channel

碩士
國立中興大學
電機工程學系所
99
Cyclic prefix (CP) is widely used to communication systems, because it is useful and robust to cancel Inter-symbol interference (ISI). CP this technique it not only can cancel ISI, but also to reduce the channel with the Channel-Shortening Equalizer (CSE). This technique CSE is using this way that the length of channel is equal with the length of CP the last CP subtract the last symbol will be zero. Follow this way if other CPs with other copy symbols subtraction also zero or minimum to zero, the channel is equal to one tap channel for the transmission data. Using this way demodulate the transmission data, we can use the sample (one tap) frequency-domain equalization to restore it. And orthogonal frequency-division multiplexing (OFDM) and Code Division Multiple Access (CDMA) is a popular transmission format for emerging wireless communication systems, including satellite radio, various wireless local area network (LAN) standards, and digital broadcast television. So in this paper we also use the CSE in the DS-CDMA and the OFDM-CDMA system.

碩士
國立中山大學
通訊工程研究所
96
Considering the communication systems with cyclic prefix (CP), such as orthogonal frequency-division multiplexing (OFDM) modulation and single-carrier cyclic prefixed (SCCP) modulation, when the length of CP is longer than the channel length, the use of cyclic prefix (CP) does not only eliminate the inter-block interference, but also convert linear convolution of the transmitted signal with the channel into circular convolution. Unfortunately, the use of CP significantly decreases the bandwidth utilization. Therefore, to reduce the length of CP is a critical issue. The thesis investigates that how to design a channel-shortening equalizer (CSE) at receiver which forces the length of the effective channel response as short as the CP length. The thesis describes the signal model as a matrix form. The effect channel response after CSE is investigated and then the coefficient of channel shortening filter is obtained using singular value decomposition method under various criterions. We further propose a novel CSE maximizing the shortening signal-to-interference ratio. In addition, it is demonstrated that the proposed CSE has the same performance as the conventional scheme but a lower computation complexity.

碩士
國立臺灣大學
電子工程學研究所
99
With the popular application of wireless local area network (WLAN), there is an increasing demand for bandwidth by the users. Some existing WLAN specifications cannot provide adequate transmission rate gradually. The main purpose of IEEE 802.11n WLAN standard is to provide a higher transmission rate to meet present and future bandwidth requirements. The difference between IEEE 802.11n and previous standards is the use of multiple input multiple output (MIMO) technique combining with OFDM which causes substantially improvement of transmission rate. In orthogonal frequency division multiplexing (OFDM) systems, a time-domain equalizer (TEQ) is used to reduce the inter-symbol interference (ISI) by shortening the channel impulse response when the channel length is larger than cyclic prefix (CP) length. However, conventional channel shortening methods may have frequency notch problem which will cause performance degradation. In this thesis, we propose a channel shortening algorithm to effectively mitigate the frequency notch effect. Besides, we also extend the proposed algorithm to MIMO environment with joint channel shortening technique. The simulation results show that the proposed algorithm has the best system performance in the MIMO-OFDM system as compared with other channel shortening algorithms. We use the delay estimate method, matrix property and Gauss-Seidel iterative method to reduce the high computation complexity TEQ design. The proposed TEQ algorithm architecture can provide SISO to MIMO environment and the one TEQ can be shared for other receivers with only twenty complex multipliers by folding technique. Finally, the TEQ engine is implemented in UMC90 40 MHz with 1.91 mm2. And it can provide the MIMO environment from 1X1 to 4X4.

碩士
國立中興大學
電機工程學系所
104
In the Fifth Generation(5G) communication system ,the research about it can be sorted higher data rates ,massive number of devices and low-latency high-reliability ,etc. The Short-Packet Communications (SPC) is a concerned application in the subject about low-latency high-reliability. Due to the property of SPC, efficiently reducing the length of cyclic prefix (CP) and length of pilot is a important study. This paper proposes a blind receiver for SPC system that do not need to joint pilot and can solve the interblock interference (IBI) caused by CP length not enough. The proposed receiver exhibits a four-stage structure: the first stage performs blind shortening of channel impulse responses (CIRs),without needing neither a priori knowledge of the CIRs to be shortened, nor preliminary compensation of the CFOs; the second stage performs joint estimation and compensation of the CFOs; the third stage implements signal-to-noise ratio (SNR) maximization, without requiring knowledge of the shortened CIRs; the fourth stage performs blind adaptive equalization with fractional lower-order constant modulus algorithm(FLOS _CMA) algorithm to update the equalization.

碩士
國立交通大學
電信工程系所
97
In order to cancel nonlinear acoustics echo in hands-free telephones or teleconferencing system. In general, adaptive Volterra filter and Hammerstein model are known to track nonlinear echo path. However, their major drawbacks are slow convergence rate and high computation complexity. In this thesis, we propose an optimum time–and tap– variant step-size for Volterra filter in order to speed up convergence rate. The step-size is based on the MMSE criterion of coefficients errors. As the optimum step-size needs to know the real echo path coefficient, we propose the exponential model for practical implementations. In addition to adaptive step-size control , the channel shortening structure was proposed to overcome slow convergence rate and high computation complexity in Hammerstein structure, we perform the least-square and adaptive algorithm theoretical analysis in channel shortening structure in case of a linear loudspeaker. From which a multiple stage update scheme is proposed in this structure to speed up convergence rate. Computer simulations justify our analysis and show the improved performance of the proposed nonlinear acoustic echo canceller.

碩士
國立清華大學
通訊工程研究所
95
UWB system is one of the most popular broadband wireless access technologies. Its main advantage is the capability of transmitting high rate data in a short distance. It is also a potential candidate for future wireless personal area network, and can be widly adopted in transmission of data, video and voice. In this thesis, we implement a baseband inner receiver for MB-OFDM UWB, including synchronization, Fast Fourier Transform, and channel equalization circuits. We also propose a channel shortening with receiver diversity scheme to deal with multipath environment. Detail simulations and derivations are also given in this thesis.

碩士
國立中山大學
通訊工程研究所
97
The Orthogonal frequency division multiplexing (OFDM) modulator with redundancy has been adopted in many wireless communication systems for higher data rate transmissions .The block transmission of signal-blocks through the channel will suffer from the inter-block interference (IBI) and inter-symbol interference (ISI). In the traditional transmitter of the OFDM systems, redundancy (or guard interval), such cyclic prefix (CP) or zero padding (ZP), with sufficient length, is inserted in the transmitted block to avoid the IBI. In this thesis, we propose a novel pseudo random cyclic postfix (PRCP-) OFDM system configuration and joint a blind channel shortening algorithm which named MERRY algorithm [18], which adopts the PRCP as redundancy and combines with multiple antennas. In fact, the multiple input and multiple output (MIMO) system, which exploits the spatial diversity, it can be used to further enhance the channel capacity and achieve high data-rate, and we extend the PRCP-OFDM to the MIMO case with space-time block coding. In redundancy insufficient case, the blind channel shortening algorithm be adopted for suppressing the IBI. The main property of PRCP-OFDM modulation is that it exploits the cyclic-postfix sequences to estimate channel information with a low complexity method. For CP-OFDM, it overcomes the channel null problem. Compared with ZP-OFDM, it uses the additional information to estimate channel which is replaced by zero samples in ZP-OFDM. Moreover, PRCP-OFDM avoids the interference of signals to the desired postfix when we estimate channel impulse response (CIR) and which is different from pseudo random postfix (PRP-) OFDM [8]. Thus, when SNR grows, PRCP-OFDM can have better performance than PRP-OFDM. With the help of [9], [12] and [13]. Via computer simulation, we verify that the performance is improved.

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