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检索条件"主题词=Linear predictive coding"

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Integrating Kalman filtering and multi-pulse coding for speech enhancement with a non-stationary model of the speech signal

Integrating Kalman filtering and multi-pulse coding for spee...

引用

Asilomar Conference on Signals, Systems & Computers

作者： Chunjian Li S.V. Andersen Department of Communication Technology University of Aalborg Aalborg Denmark

In this paper, speech enhancement via Kalman filtering is considered. A non-stationary signal model for the speech signal is first described. This model consists of a slowly varying AR model and an excitation source that exhibits a rapidly time-varying variance. The AR model and the excitation model fit nicely into the Kalman filtering framework, fully exploiting the capability of the Kalman filter to process non-stationary signals in an LMMSE optimum manner. The AR-model coefficients are estimated by a decision-directed type power spectral subtraction method followed by an LPC analysis. For the robust estimation of the rapidly time-varying excitation model in the presence of noise, we propose the use of a multi-pulse linear predictive coding (MPLPC) based method. The Kalman filtering algorithm based on the non-stationary signal model is able to partially avoid the commonly used quasi-stationarity assumption of the speech. Therefore the non-stationarity of the signal is fully exploited in suppressing the noise power that is more stationary. Our experiments show that the Kalman filter with rapidly time-varying variance modeling using the proposed MPLPC based method brings significant performance improvement both when compared to a baseline Kalman filtering method with quasi-stationarity assumption and when compared to the well-known MMSE log-spectral amplitude estimator (MMSE-LSA).

关键词： Speech enhancement Kalman filters Filtering Speech coding Wiener filter Signal resolution Signal processing linear predictive coding Noise robustness Speech analysis

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Process monitoring in principal component subspace: part 2. Fault identification and isolation study

Process monitoring in principal component subspace: part 2. ...

引用

IEEE International Conference on Systems, Man and Cybernetics

作者： Ning Jiang Haiqing Wang Diancai Yang Institute of Process Equipment & Control Engineering Zhejiang university of Technology Hangzhou China National Laboratory of Industrial Control Technology Institute of Industrial Process Control University of Zhejiang Hangzhou China Qingdao Mesnac Company Limited Qingdao China

ISBN: (纸本)0780385675

Principal component analysis (PCA) finds wide applications in process monitoring and product quality control. The objectives of this part of the two-part paper are (i) development of fault identification method based on the fault reconstruction results obtained in part 1 paper and (ii) analyses the fault isolatability issue to ascertain if a fault could be uniquely identified. Both the aims are restricted in the principal component subspace (PCS) as in part 1. It is shown the isolatability of the candidate fault set can be explored in prior using only normal process data. Then a reconstruction based faults identification index is proposed. Again the obtained results are illustrated and verified by simulation studies on a double-effective evaporator.

关键词： Monitoring Fault diagnosis Principal component analysis linear predictive coding Industrial control Redundancy Isolation technology Process control Quality control Personal communication networks

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Reducing computational and memory cost for cellular phone embedded speech recognition system

Reducing computational and memory cost for cellular phone em...

引用

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

作者： C. Levy G. Linares P. Nocera J.-F. Bonastre Laboratoire Informatique dAvignon Avignon France

We present several methods able to fit speech recognition system requirements to cellular phone resources. The proposed techniques are evaluated on a digit recognition task using both French and English corpora. We investigate particularly three aspects of speech processing: acoustic parameterization, recognition algorithms; acoustic modeling. Several parameterization algorithms (LPCC, MFCC and PLP) are compared to the linear predictive coding (LPC) included in the GSM norm. The MFCC and PLP parameterization algorithms perform significantly better than the others. Moreover, feature vector size can be reduced to 6 PLP coefficients, allowing memory and computation resources to be decreased without a significant loss of performance. In order to achieve good performance with reasonable resource needs, we develop several methods to embed a classical HMM-based speech recognition system in a cellular phone. We first propose an automatic on-line building of a phonetic lexicon which allows a minimal but unlimited lexicon. Then we reduce the HMM complexity by decreasing the number of (Gaussian) components per state. Finally, we evaluate our propositions by comparing dynamic time warping (DTW) with our HMM system - in the cellular phone context - for clean conditions. The experiments show that our HMM system outperforms DTW for speaker independent tasks and allows more practical applications for the cellular-phone user interface.

关键词： Embedded computing Computational efficiency Cellular phones Speech recognition Hidden Markov models Mel frequency cepstral coefficient linear predictive coding Speech processing GSM Vectors

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Formant frequency estimation in noise

Formant frequency estimation in noise

引用

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

作者： Bin Chen P.C. Loizou Department of Electrical Engineering University of Texas Dallas Richardson TX USA

This paper addresses the problem of formant frequency estimation of speech signals corrupted by colored noise. The spectrum is sequentially segmented into K segments so that each segment contains a single formant. A segmentation metric based on Wiener filter theory is proposed for determining the segment boundaries. A peak-picking algorithm is used for estimating the formant frequencies in each segment. Results obtained using vowels embedded in +5 dB S/N speech-shaped noise, indicated that the proposed algorithm produced formant frequencies which were comparable to those estimated in quiet.

关键词： Frequency estimation Noise level Speech enhancement Wiener filter linear predictive coding Resonator filters Optimized production technology Colored noise Speech synthesis Dynamic programming

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Code excited linear prediction codec for electrocardiogram

Code excited linear prediction codec for electrocardiogram

引用

Annual International Conference of the IEEE Engineering in Medicine and Biology Society (EMBC)

作者： S. Banik R.J. Martis D. Nayak Electronics & Communication Engineering MIT Manipal India Department of Electronics & Communication Engineering MIT Manipal India

In This work we propose a CELP ECG codec for medical telemetry. The encoding algorithm is based on code-excited linear prediction (CELP). The general framework proposed is: QRS detection, calculation of LPC parameter, generation of residual error signal, codebook generation, MSE (mean square error) search. The codebook is generated for residual error. The indices of the codebook and corresponding LPC parameters are transmitted where the minimum MSE occurs. A replica of the transmitter codebook is present at the receiver. Corresponding to the received index value residual error coefficients are retrieved from the receiver codebook. The ECG signal is reconstructed from the retrieved code word.

关键词： Codecs Electrocardiography Filters Signal synthesis Biomedical engineering Medical diagnostic imaging linear predictive coding Biomedical monitoring Signal generators Bandwidth

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Investigation on differences between whispered and phonated sustained Turkish vowels

Investigation on differences between whispered and phonated ...

引用

IEEE Signal Processing and Communications Applications (SIU)

作者： F. Icat H.G. Ilk Ankara Üniversitesi Ankara Turkey

Whispering is a special speaking model where phonated speech is reduced. This model of speaking is used for various reasons in daily life. Whispered and phonated Turkish sustained /a/, /e/, /i/, /o/, /u/ vowels are investigated. Time domain, spectrum, energy, formant frequencies and bandwidth differences between these vowels are investigated for whispered and phonated speech.

关键词： Speech linear predictive coding Frequency Mathematical model Bandwidth

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Harmonicity and dynamics-based features for audio

Harmonicity and dynamics-based features for audio

引用

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

作者： H. Srinivasan M. Kankanhalli Applied Research Group Satyam Computer Services Ltd Bangalore Department of Computer Science National University of Singapore Singapore

Features are very important for audio processing. Tasks like speech recognition and instrument identification are based on features. Most low-level features currently used are based on LPC and cepstral analysis. We propose a class of features based on dynamics and harmonicity. In particular, we define the notion of harmonic derivative. The efficacy of the features is demonstrated for music genre classification and instrument family classification. In particular, the features are shown to be cepstrum-equivalent.

关键词： Image analysis Frequency Speech recognition Instruments Cepstral analysis linear predictive coding Ear Layout Computer science Information analysis

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Speaker recognition based on transformed line spectral frequencies

Speaker recognition based on transformed line spectral frequ...

引用

IEEE International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)

作者： Bong Jin Lee S. Kim Hong-Goo Kang MCSP Yonsei University Seoul South Korea

Line spectral frequencies (LSF) and five types of transformed LSF are studied for robust text-independent speaker identification. Transformations are constructed by considering physical aspects of the vocal tract. These aspects are: location of formants/nulls; bandwidth of formants/nulls; bandwidth and location of formants; bandwidth and location of nulls; interval of adjacent formant and null locations. Identification tests using the TIMIT database verify that all features are useful for speaker recognition; the bandwidth and location of formants, especially, show the best performance. Simulation results also show that LSF and some of the transformed LSF give better performance than Mel-frequency cepstral coefficient (MFCC).

关键词： Speaker recognition Bandwidth Mel frequency cepstral coefficient Spatial databases Polynomials Equations Robustness linear predictive coding Testing Cepstral analysis

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A model-based clustering for time-series with irregular interval

A model-based clustering for time-series with irregular inte...

引用

International Conference on Machine Learning and Cybernetics (ICMLC)

作者： Xiao-Tao Zhang Wei Zhang Xiong Xiong Qi-Wen Wang Cui-Yu Li School of Management Tianjin University Tianjin China Tianjin University of Finance & Economics Tianjin China School of Mechanical Engineering Tianjin University Tianjin China

ISBN: (纸本)0780384032

Clustering problems are central to many knowledge discovery and data mining tasks. However, most existing clustering methods can only work with fixed-interval representations of data patterns, ignoring the variance of time axis. In this paper, we study the clustering of data patterns that are sample in irregular interval. In the paper, a model-based approach using cepstrum distance metrics and autoregressive conditional duration (ACD) model is proposed. Experimental results on real datasets show that this method is generally effective in clustering irregular space time series, and conclusion inferred from experimental results agrees with the market microstructure theories.

关键词： Hidden Markov models Data mining Brain modeling Gene expression linear predictive coding Microstructure Cepstral analysis Speech analysis Time series analysis Electroencephalography

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Wideband audio over narrowband low-resolution media

Wideband audio over narrowband low-resolution media

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

作者： Heping Ding Institute for Microstructural Sciences National Research Council Canada

ISBN: (纸本)0780384849

This paper proposes a scheme of transmitting or storing a wideband audio stream in a channel or storage medium which is narrowband in nature and with a low resolution data format, such as an 8-bit companded one. In the proposed scheme, signal components in the upper-band part of the wideband beyond the narrowband - are encoded and then the code is embedded into the narrowband stream in a least audible way. As a result, this scheme enables the transmission and storage of wideband audio signals with the existing narrowband infrastructure. Furthermore, the scheme is fully backward compatible, i.e., conventional receivers, without the proposed decoding mechanism, can still access the narrowband audio with little audible degradation. The proposed scheme can find applications in areas where a high quality wideband audio is needed but the physical bandwidth is limited. Examples are voice-over-IP and telephony in general, as well as digital storage and playback.

关键词： Wideband Narrowband Niobium Payloads Frequency Transmitters Sampling methods linear predictive coding National security Decoding

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