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Algebraic codebook search strategy for an algebraic code-excited linear-prediction speech coder by means of reduced candidate mechanism and iteration-free pulse replacement

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IET SIGNAL PROCESSING 2014年第9期8卷 990-995页

作者： Yeh, Cheng-Yu Natl Chin Yi Univ Technol Dept Elect Engn Taichung 41170 Taiwan

This work aims to present a combined version of reduced candidate mechanism (RCM) and iteration-free pulse replacement (IFPR) as a novel and efficient way to enhance the performance of algebraic codebook search in an algebraic code-excited linear-prediction speech coder. As the first step, individual pulse contribution in each track is given by RCM, and the value of N is then specified. Subsequently, the replacement of a pulse is performed through the search over the sorted top N pulses by IFPR, and those of 2-4 pulses are carried out by a standard IFPR. Implemented on a G.729A speech codec, this proposal requires as few as 20 searches, a search load tantamount to 6.25% of G.729A, 31.25% of the global pulse replacement method (iteration = 2), 41.67% of IFPR, but still provides a comparable speech quality in any case. The aim of significant search performance improvement is hence achieved in this work.

关键词： search problems vocoders speech coding iterative decoding algebraic codes linear predictive coding speech codecs algebraic codebook search strategy algebraic code excited linear prediction speech coder reduced candidate mechanism iteration free pulse replacement RCM IFPR standard IFPR G.729A speech codec global pulse replacement method search performance improvement

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Two-stage optimisation algorithm for adaptive IIR notch filter

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ELECTRONICS LETTERS 2014年第14期50卷 985-U96页

作者： Peng, Renhua Zheng, Chengshi Li, Xiaodong Chinese Acad Sci Commun Acoust Lab Inst Acoust Beijing 100190 Peoples R China Chinese Acad Sci Shanghai Adv Res Inst Acoust & Informat Technol Lab Beijing 100190 Peoples R China

The performances of the linear prediction (LP) algorithm and the indirect plain gradient (IPG) algorithm for the adaptive infinite-impulse-response (IIR) notch filter are theoretically studied, and the two phenomena in these two algorithms are revealed. First, the LP algorithm has a faster convergence speed than the IPG algorithm when the initial adjusted value is far away from the optimal solution. Secondly, the IPG algorithm has a much smaller estimation bias and mean squared error than the LP algorithm in a steady state. On the basis of these facts, a two-stage optimisation algorithm is proposed to make full use of the advantages of both the LP algorithm and the IPG algorithm. Simulation results show that the proposed algorithm can improve the convergence rate while keeping the same steady state as the IPG algorithm.

关键词： IMPULSE response MATHEMATICAL optimization STOCHASTIC convergence ALGORITHMS linear predictive coding

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Digital linear pre-compensation technique to enhance predistortion performance in multicarrier DVB-S2 satellite communication systems

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ELECTRONICS LETTERS 2014年第13期50卷 957-958页

作者： Allegue-Martinez, M. Kelly, N. Zhu, A. UCD Sch Elect Elect & Commun Engn RF & Microwave Res Grp Dublin Ireland

Nonlinearity in communication satellite payloads often severely degrades the downlink signal quality in such systems. A digital pre-compensation technique to enhance digital predistortion performance in multicarrier satellite communication systems is presented. The proposed technique has been validated by simulations with an appropriate platform where excellent performance improvements have been measured. In addition, the implementation feasibility of this technique has been briefly demonstrated by means of a real-time FPGA implementation.

关键词： ARTIFICIAL satellites in telecommunication TELECOMMUNICATION systems COMMUNICATION REMOTE broadcasting linear predictive coding

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Signal detection assisted from channel weight prediction for delay-sensitive orthogonal frequency-division multiplexing communications

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IET COMMUNICATIONS 2014年第10期8卷 1745-1750页

作者： Lv, Changwei Lin, Jia-Chin Hou, Shujuan Mei, Wenbo Beijing Inst Technol Sch Informat & Elect Beijing 100081 Peoples R China Natl Cent Univ Dept Commun Engn Jhongli Taiwan Chongqing Commun Inst Dept Informat Technol Chongqing 400035 Peoples R China

In signal coherent detection, data-aided (DA) method has high detection accuracy, but unavoidably suffers from a large processing delay. Meanwhile, a decision-directed (DD) method incurs no processing delay, but suffers from error propagation problem. Motivated by the above problems of DA and DD methods, the authors present a novel signal detection algorithm for delay-sensitive applications employing orthogonal frequency-division multiplexing systems. The proposed detection method provides similar detection accuracy to the DA method and incurs no processing delay. This method employs channel estimation followed by linear prediction and interpolation to obtain the channel-state information for symbol detection. To achieve a better prediction performance, they perform the linear prediction on the channel taps in the time domain, and conduct the interpolation using a raised cosine filter. Simulations on physical channel models verify the improvements of the proposed method.

关键词： SIGNAL detection ORTHOGONAL frequency division multiplexing linear predictive coding INTERPOLATION COSINE function

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Adaptive Total Least Squares Based Speech Prediction 6

Adaptive Total Least Squares Based Speech Prediction

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6th International Conference on Information Technology and Multimedia (ICIM)

作者： Javed, Shazia Ahmad, Noor Atinah Univ Sains Malaysia Sch Math Sci George Town 11800 Malaysia

ISBN: (纸本)9781479954230

In this paper, an instantaneous total error based adaptive linear predictor is presented for linear predictive coding (LPC) of speech signals. In LPC, the speech signal is predicted by a linear combination of delayed input signals that are contaminated by noise. For this reason, total least mean squares (T-LMS) algorithm is used to decode the noisy input signals and to predict a speech signal. A compressed speech prediction is done when the mean squares total error is minimized, showing the efficiency of T-LMS based LPC model. Experimental results are recorded for different values of signal to noise ratio (SNR) of the input signals, and a comparative study is presented with instantaneous error squares based adaptive filter. These results show the preference of proposed predictor over the other.

关键词： linear predictive coding adaptive filter total least squares

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linear Prediction Analysis of Crosscorrelation Sequence for Voiced Speech

Linear Prediction Analysis of Crosscorrelation Sequence for ...

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Annual Summit and Conference of Asia-Pacific-Signal-and-Information-Processing-Association (APSIPA)

作者： Liu, Liqing Shimamura, Tetsuya Saitama Univ Saitama Japan

ISBN: (纸本)9786163618238

The conventional linear prediction (LP) analysis is known to suffer from problems that it is sensitive to additive noise. In this paper a new approach for LP analysis of crosscorrelation sequence between speech signal and its zero-crossing wave has been presented. Simulation results show that the proposed method is capable of performing the speech analysis under a white noisy environment.

关键词： correlation methods linear predictive coding speech processing LP analysis additive noise crosscorrelation sequence linear prediction analysis speech analysis speech signal voiced speech white noisy environment zero-crossing wave Additive noise Cepstrum Correlation Noise measurement Signal to noise ratio Speech

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Feature Enhancement for Classifier Optimization and Dimensionality Reduction Speech Pathology Detection - A case study 11

Feature Enhancement for Classifier Optimization and Dimensio...

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11th Annual IEEE India Conference (INDICON)

作者： Shilaskar, Swati Ghatol, Ashok Govt Coll Engn Dept Elect & Telecommun Amravati 444604 India KJs Educ Inst Pune 411048 Maharashtra India

ISBN: (纸本)9781479953646

Voice is important for professionals like speakers, teachers, actors, singers and it is the important tool for communication. Laryngeal pathologies induce perturbations in the speech signal. Speech signal is discriminated as pathological or healthy based on roughness - breathiness - hoarseness (RBH) in the quality of signal. In recent years pattern recognition along with various signal processing techniques has emerged as an effective non invasive tool for diagnosis of pathological condition. Signal processing techniques tend to generate large number of features representing the signal. Automatic feature reduction techniques are vital in identifying the relevant features and eliminating the redundant ones. We extract features from speech signal using the acoustic analysis. Features are enhanced by alleviating gender bias. Periodic variations in the signal are captured using statistical techniques. We investigate intelligent system to generate reduced feature subset with improvement in diagnostic performance.

关键词： Acoustic Cepstral Feature extraction linear predictive coding Mel Pathology Spectral Voice quality

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Back Propagation Feed forward neural network approach for Speech Recognition 3

Back Propagation Feed forward neural network approach for Sp...

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International Conference on Reliability, Infocom Technologies and Optimization (ICRITO) (Trends and Future Directions)

作者： Rajput, Neelima Verma, S. K. GBPEC Dept CSE Pauri Gharwal Uttrakhand India

ISBN: (纸本)9781479968961

In this paper a biologically motivated approach for the English alphabet speech recognition is implemented by using a self-organized neural network. The designing of an accurate and effective speech recognition system is a challenging task in the area of human computer interface. linear predictive coding (LPC) is used for learn Feature extraction of input audio signals. Back propagation (BP) is a feed forward neural network and it propagates the error in backward direction to update the weights of hidden layers. The error is difference of actual output and target output computed on the basis of gradient descent method. The performance of the system is evaluated on the basis of recognition rate. We have used BP neural network architecture to recognize the time varying input data. The proposed provides better accurate results than the existing systems for the English Alphabet speech recognition.

关键词： Artificial Neural Networks Back Propagation Feed Forward Neural Network Gradient Descent Method linear predictive coding Speech Recognition System

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Application of DSP Speech Synthesis System on Service Robots

Application of DSP Speech Synthesis System on Service Robots

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CACS International Automatic Control Conference (CACS 2014)

作者： Huang, Guo-Shing Lu, Yan-An Natl Chin Yi Univ Technol Inst Elect Engn Taichung Taiwan

ISBN: (纸本)9781479945849

The main purpose of this paper is to perform the voice signal processing and synthesis to apply to service robots using DSP TMS320C6713. In this study, we select the C language program written with DSP;the voice signal is restored to play. The speech synthesis is implemented using linear predictive coding (LPC) approach in the paper. Because the LPC synthesis is a coding technique of time waveform, one can reduce the transmission rate of signal in time domain;save perfect voice messages. It can be obtained a very high naturalness and clarity for the synthesis of a group of words. We scored the synthesis results by subjective listening test ways.

关键词： DSP speech recognition service robots linear predictive coding speech synthesis

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ERROR RESILIENCE ENHANCEMENT FOR A ROBUST ADPCM AUDIO coding SCHEME

ERROR RESILIENCE ENHANCEMENT FOR A ROBUST ADPCM AUDIO CODING...

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IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

作者： Simkus, Gediminas Holters, Martin Zoelzer, Udo Helmut Schmidt Univ Univ Fed Armed Forces Dept Signal Proc & Commun Hamburg Germany

ISBN: (纸本)9781479928934

This work comprises an extension of a backward adaptive quantizer which is employed together with a robust lattice predictor in an ADPCM coding scheme. Predictors of the ADPCM audio coding schemes are often considered as the part most sensitive to transmission errors. Nevertheless, a single transmission error causes a short destabilization of the adaptive quantizer at the decoder side. Therefore, this destabilization boosts the deviation of the prediction filter at the decoder from the encoder side. Desired damping of the quantizer shortens synchronization periods. However, damping leads to degradation of the quantizer's adaptation properties and consequently to a decrease in audio quality. We show that the transmission of the quantizer's envelope to the decoder in short intervals and in combination with envelope error detector reduces the impairment of the reconstructed audio signal in noisy transmissions. An objective audio quality evaluation confirms significant quality improvement at BER higher than 10(-4) and no quality degradation if an ideal channel is employed.

关键词： Audio coding error robust transmission linear predictive coding robust ADPCM

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