Clarity and intelligibility in speech signal demands removal of noise and interference associated with the signal at the source. This poses further challenge when the speech signal is colored with human emotions. In t...
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ISBN:
(纸本)9781509012770
Clarity and intelligibility in speech signal demands removal of noise and interference associated with the signal at the source. This poses further challenge when the speech signal is colored with human emotions. In this work, the authors have taken a novel step to enhance the emotional speech signal adaptively before classification. Most popular adaptive algorithm such as Least mean square (LMS), Normalized least mean squares (NLMS) and Recursive least square (RLS) has been put to test to obtain enhanced speech emotions. Neural network based Multilayer perceptron (MLP) classifier is used to recognize fear speech emotion as against neutral voices using effective Linear Prediction coefficients (LPCs). The accuracy has improved to approximately 77% with enhanced signal. The increased accuracy of this signal has been witnessed with RLS algorithm as against the noisy signal with corresponding algorithm.
A novel adaptation strategy is proposed for Acoustic Echo Cancellation (AEC). The new algorithm firstly partitions the adaptive filter into several blocks and the successive blocks with the maximum l(2) norm are consi...
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ISBN:
(纸本)9781538631423
A novel adaptation strategy is proposed for Acoustic Echo Cancellation (AEC). The new algorithm firstly partitions the adaptive filter into several blocks and the successive blocks with the maximum l(2) norm are considered to be the active blocks. Then the coefficients of the active blocks are adapted with large update probability to ensure the identification accuracy while the zero coefficients are adapted with small update probability to decrease the number of adapting coefficients. By excluding most of the zero coefficients from adaptation, the new algorithm improves its performance in terms of convergence and computation. The simulation performed in the context of AEC demonstrates the robustness and the advantages of the new algorithm.
A fast adaptive method to eliminate a single sinusoidal interference of known frequency is proposed. The coupled-oscillator is used to generate the reference signal instead of look-up table. Unlike the least mean squa...
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ISBN:
(纸本)9781538623176
A fast adaptive method to eliminate a single sinusoidal interference of known frequency is proposed. The coupled-oscillator is used to generate the reference signal instead of look-up table. Unlike the least mean square (LMS) based adaptive algorithm, the weighted least squared error criterion is employed instead to evaluate the system parameters. It has been shown that the proposed method can quickly eliminate unwanted interference as compared to previous methods. Moreover, the extensive simulations have been done and revealed that proposed algorithm yields a constant output signal to interference ratio (SIR) of about 20 dB regardless of input SIR.
Traditional monitoring system of curing machines can hardly meet controlling requirement not only because it is interfered easily by the hostile environment of locale in the respect of data acquisition, but also it ha...
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ISBN:
(纸本)9781424427239
Traditional monitoring system of curing machines can hardly meet controlling requirement not only because it is interfered easily by the hostile environment of locale in the respect of data acquisition, but also it has a low processing speed of data and weak communication ability. This study establishes a CAN bus monitoring network which can fully utilize the powerful digital signal processing ability of DSP, and adopts a new sort of variable step LMS adaptive filtering algorithm. The algorithm resolves better the contradiction between filtering speed and precision, and then improves the accuracy of data acquisition and the real-time data processing through controlling step-size factor. Furthermore, it optimizes the performance of the filter. The CAN bus network achieve effectively the real-time sharing and monitoring of data, besides, because of the open structure, the study also supplies a foresighted thought for the later construction and capacity expansion of the system.
In this paper,we presented the derivation of an adaptive algorithm for time-varying system and verified that it satisfied convergence *** on the concept of distance in parameter space,we proposed a robust identificati...
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ISBN:
(数字)9789881563934
ISBN:
(纸本)9781538629185
In this paper,we presented the derivation of an adaptive algorithm for time-varying system and verified that it satisfied convergence *** on the concept of distance in parameter space,we proposed a robust identification method which identified the smallest set of discrete transfer functions under the assumption that its parameters were time-varying and described derivation of an adaptive law based on verification and update algorithm of the identified model *** also described that the proposed algorithm satisfied the convergence properties through numerical simulation.
A new algorithm for adaptive processes is presented here. The discussion is in the context of a particular MOS LSI sampled analog realization, but the applicability is more general. The algorithm implemented is a simp...
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In this paper a module consisting of a Fast Least Mean Square (FLMS) filter is modeled and verified to eliminate acoustic noise, which is a problem in voice communication. However the acoustic noise cancellation (ANC)...
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ISBN:
(纸本)9781479935666
In this paper a module consisting of a Fast Least Mean Square (FLMS) filter is modeled and verified to eliminate acoustic noise, which is a problem in voice communication. However the acoustic noise cancellation (ANC) is modeled using digital signal processing technique especially Simulink Blocksets. The needed algorithm code is generated in Matlab Simulink programming. At the simulation level, results of simulink implementation prove the module behavior for cancellation of noise in voice communication using the FLMS adaptive algorithm. The main scope of this paper is to implement the module, benefiting the advantage of DFT (Discrete Fourier transform) circular convolution properties and Fast Fourier Transform (FFT) high computation speed in frequency domain rather adaptive algorithms Normalized Least Mean Square (NLMS) and Recursive Least Square (RLS) in time domain with high complexity, also the simplicity of the implementation using simulink programming.
An adaptive algorithm is proposed to find multiple roots of polynomials which were not well solved by the other methods. Its convergence was presented and proved. The computation is carried out by simple steepest desc...
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ISBN:
(纸本)9783540877332
An adaptive algorithm is proposed to find multiple roots of polynomials which were not well solved by the other methods. Its convergence was presented and proved. The computation is carried out by simple steepest descent rule with adaptive variable learning rate. The specific examples showed that the proposed method can find the multiple roots of polynomials at a very rapid convergence and very high accuracy with less computation.
For a communications channel with the two states ("Good" and "Bad"), we examine the simulation model of the communications channel state detection adaptive algorithm with a fixed period of observat...
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ISBN:
(纸本)5778204396
For a communications channel with the two states ("Good" and "Bad"), we examine the simulation model of the communications channel state detection adaptive algorithm with a fixed period of observation.
This paper investigates multimedia quality fairness in wireless LAN environments where channel are error-prone due to mobility and fading. The experimental results show that using fixed MAC arguments for nodes in hete...
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ISBN:
(纸本)9781424403660
This paper investigates multimedia quality fairness in wireless LAN environments where channel are error-prone due to mobility and fading. The experimental results show that using fixed MAC arguments for nodes in heterogeneous channel conditions leads to unequal throughput performance and that may incur the degradation of multimedia QoS. To overcome the unfairness problem for provisioning QoS, we propose a cross-layer adaptation scheme by on-line adapting the multidimensional MAC-layer backoff parameters depending on the application-layer QoS requirements and PHY-layer channel conditions. Our solution is based on an optimization approach which utilizes neural networks to learn the cross-layer function. Simulation results demonstrate that our adaptive scheme can tackle a variety of channel condition to provide fair throughput for nodes in heterogeneous channel conditions.
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