“…The signal power is then given by (8) Next, we compute the noise power by considering only the noise component of (5). In this case, (6) becomes (9) where .…”
Section: B Additive White Gaussian Noise Casementioning
confidence: 99%
“…Subband adaptive filtering has been used in applications such as acoustic echo cancellation (AEC) in order to address the problems of slow convergence and high computational complexity associated with long adaptive filters [5]. One approach is to divide the fullband signal into multiple, lower rate subband signals that interfere (alias) as little as possible with each other.…”
Abstract-Adaptive Direct Sequence Spread Spectrum (DSSS) receivers have advantages over their fixed matched filter counterparts including interference cancellation capabilities and simplification of PN code acquisition. However, convergence using the LMS algorithm will be very slow in situations with relatively high SNR and/or a large number of users. The use of the RLS algorithm will improve convergence speed but at significantly increased computational cost, especially for long PN codes. Unfortunately, computationally efficient, fast RLS algorithms cannot be used because the filter is updated at the symbol rate rather than at every sample. In this paper, we propose a subband version of the RLS-based receiver that utilizes multiple, shorter length adaptive filters. This approach significantly reduces computation and introduces architectural parallelism into the system implementation. We design an optimal subband transform and provide simulation results demonstrating the improved convergence properties as compared with the fullband system. Index Terms-Adaptive direct sequence spread spectrum, parallel receiver, subband transforms.
“…The signal power is then given by (8) Next, we compute the noise power by considering only the noise component of (5). In this case, (6) becomes (9) where .…”
Section: B Additive White Gaussian Noise Casementioning
confidence: 99%
“…Subband adaptive filtering has been used in applications such as acoustic echo cancellation (AEC) in order to address the problems of slow convergence and high computational complexity associated with long adaptive filters [5]. One approach is to divide the fullband signal into multiple, lower rate subband signals that interfere (alias) as little as possible with each other.…”
Abstract-Adaptive Direct Sequence Spread Spectrum (DSSS) receivers have advantages over their fixed matched filter counterparts including interference cancellation capabilities and simplification of PN code acquisition. However, convergence using the LMS algorithm will be very slow in situations with relatively high SNR and/or a large number of users. The use of the RLS algorithm will improve convergence speed but at significantly increased computational cost, especially for long PN codes. Unfortunately, computationally efficient, fast RLS algorithms cannot be used because the filter is updated at the symbol rate rather than at every sample. In this paper, we propose a subband version of the RLS-based receiver that utilizes multiple, shorter length adaptive filters. This approach significantly reduces computation and introduces architectural parallelism into the system implementation. We design an optimal subband transform and provide simulation results demonstrating the improved convergence properties as compared with the fullband system. Index Terms-Adaptive direct sequence spread spectrum, parallel receiver, subband transforms.
“…[4][5] 그러나 기존의 부밴드 적응 필터는 소위 "band-edge effect"로 인하여 수렴 속도를 향상시키는 데 한계가 있었다. [5] 이러한 band-edge effect 문제를 해결하기 위하여 minimum disturbance 원리에 기반한 정규 부밴드 적응 필터(Normalized SAF, NSAF)가 제 안되었다.…”
In this paper, we propose a variable step size algorithm to enhance the normalized subband adaptive filter which has been proposed to improve the convergence characteristics of the conventional full band adaptive filter. The well-known Kwong's variable step size algorithm is simple, but shows better performance than that of the fixed step size algorithm. However, in case that large additive noise is present, the performance of Kwong's algorithm is getting deteriorated in proportion to the amount of the additive noise. We devised a variable step size algorithm which does not depend on the amount of additive noise by exploiting a normalized adaptation error which is the error subtracted and normalized by the estimated additive noise. We carried out a performance comparison of the proposed algorithm with other algorithms using a system identification model. It is shown that the proposed algorithm presents good convergence characteristics under both stationary and non-stationary environments.
“…Among the advantages of the frequency channelized receivers compared to the more conventional time channelized (i.e., time-interleaved ADC) receivers are the ease of designing the sample/hold circuitries, greater robustness to jitter/phase noise, and reduced ADC dynamic range requirements. The main drawback of the frequency channelized receiver, however, is the slow convergence speed, which can be problematic in time-varying UWB wireless environment [3].…”
Abstract. The frequency channelized receiver enables the use of practical analog-to-digital converters (ADC) to digitize ultra-wideband (UWB) signals. The design issues of the analog and digital baseband processor for the channelized receiver in a UWB transmitted reference (TR) system are investigated. In the analog part, the receiver performance is shown to be weakly dependent on the analog filter bandwidth, the filter order, and the ADC oversampling ratio assuming white input noise. In the digital part, the coarse acquisition performance is shown to be significantly better in a channelized receiver than in a fullband receiver. The implementation issues for fine synchronization and correlation window length are also studied.
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