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检索条件"主题词=linear predictive coding"

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F0 estimation of noisy speech based on complex speech analysis

F0 estimation of noisy speech based on complex speech analys...

引用

IEEE Digital Signal Processing Workshop

作者： Tatsuhiko Kinjo Keiichi Funaki Graduate School of Science and Engineering University of Ryukyus Okinawa Japan Computing & Networking Center University of Ryukyus Okinawa Japan

ISBN: (纸本)1424405343

This paper proposes a novel robust fundamental frequency (F0) estimation algorithm based on complex-valued speech analysis for an analytic speech signal. Since analytic signal provides spectra only over positive frequencies, spectra can be accurately estimated in low frequencies. Consequently, it is considered that F0 estimation using the residual signal extracted by complex-valued speech analysis can perform better for F0 estimation than that for the residual signal extracted by conventional real-valued LPC analysis. In this paper, the autocorrelation function weighted by AMDF is adopted for the F0 estimation criterion and four signals; speech signal, analytic speech signal, LPC residual and complex LPC residual, are evaluated for the F0 estimation. Speech signals used in the experiments were corrupted by adding white Gaussian noise whose noise levels are 10, 5, 0, -5 [dB]. The experimental results demonstrate that the proposed algorithm based on complex speech analysis can perform better than other methods in an extremely noisy environment

关键词： Speech analysis Frequency estimation Signal analysis linear predictive coding Low-frequency noise Robustness Algorithm design and analysis Performance analysis Autocorrelation Speech enhancement

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Voice over IP with JVOIPLIB and JRTPLIB

Voice over IP with JVOIPLIB and JRTPLIB

引用

Conference on Local Computer Networks (LCN)

作者： J. Liesenborgs W. Lamotte F. Van Reeth Expertisecentrum Digitale Media Limburgs Universitair Centrum Diepenbeek Belgium

ISBN: (纸本)0769513212

Voice over IP (VoIP) can be used in a wide variety of applications, all having different requirements. We present JVOIPLIB and JRTPLIB, a VoIP library and an RTP library respectively. Together they make it possible to easily add VoIP to various types of applications. Both libraries are written in an object-oriented style in C++, are open-source and are both very extensible. Several measures have been taken to allow good synchronization between the communicating parties.

关键词： Internet telephony Virtual environment Delay linear predictive coding Software libraries Open source software Speech processing Signal processing Jitter Degradation

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Promising concepts and technologies for future power delivery systems

Promising concepts and technologies for future power deliver...

引用

International Universities Power Engineering Conference (UPEC)

作者： Phuong H. Nguyen Wil L. Kling Johanna M. A. Myrzik Eindhovan University of Technology Netherlands Delft University of Technnology Netherlands

The transition from a conventional delivery system to a suitable robust distribution system is emerging to adapt with different possible scenarios of future development. This paper gives a particular vision of future grid with its main requirements. An investigation of suitable concepts and technologies which draw out attentions at the present has been carried out. They are discussed regarding mentioned requirements of sustainability, efficiency, flexibility and intelligence. Active network is then introduced as the backbone of the future power delivery system. Besides, multi- agent system (MAS) is described as a potential technology to cope with anticipated challenges of future grid operation. The research described is under the framework of the Electricity Infrastructure of the Future (EIT) project carried out by cooperation of TU/e, KEMA and ECN (***).

关键词： Distributed control Supply and demand Robustness Multiagent systems Control systems Voltage control Power system stability Communication system control linear predictive coding Spine

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A comparison of SPOX and C implementations of a LPC vocoder on the TMS320C30 EVM

A comparison of SPOX and C implementations of a LPC vocoder ...

引用

IEEE Southeastcon

作者： D.A. Kilpatrick T.C. Jannett Department of Electrical and Computer Engineering University of Alabama Birmingham Birmingham AL USA

The paper describes a comparison of a C implementation of a linear predictive voice coder (LPC) and an implementation based on Spectron Microsystem's Signal Processing Operating System (SPOX). The hardware platform was a Texas Instruments TMS320C30 Evaluation Module. The SPOX and C implementations were compared based on execution time, ease of program development and maintenance, and portability to different hardware platforms. The vocoder algorithms and the results of the comparison of both implementations are presented.

关键词： linear predictive coding Vocoders Speech synthesis Speech analysis Filters Hardware Signal synthesis Synthesizers Humans Signal processing algorithms

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Comparisons of maximum system lifetime in diverse scenarios for body sensor networks

Comparisons of maximum system lifetime in diverse scenarios ...

引用

International Conference on Computer and Information Technology (ICCIT)

作者： Nazrul Islam Mondal Shahid Uz Zaman Abdullah Al Masud Jahangir Alam Department of Computer Science & Engineering Rajshahi University of Engineering & Technology Bangladesh Department of Electronic and Telecommunication International Islamic University Chittagong Bangladesh Department of Computer Science & Engineering Rajshahi University of Engineering and Technology Bangladesh Department of Electronic and Telecommunication International Islamic University Chittagong Bangladesh

In this paper, the main target is to compare the system lifetimes in diverse scenarios based on our new energy dissipation model in the body sensor network as well as maximize the system lifetime of sensor nodes when they will make communication among body sensors and personal communication unit. Nowadays, the ultra low energy consumption is the very much important challenge for medical applications. The best compression technique like LPC is selected for energy saving based on some calculations. This research work studies and analyzes the transceiver energy consumption for different compression algorithms and select the best technique like LPC for energy saving. Formulation of a linear programming problem is also the important part of this research work, where is to maximize the system lifetime which is equivalent to the time until the first node runs out of battery. Maximum system lifetimes are calculated by MATLAB optimization technique using and without using efficient compression algorithm like LPC in various environments. Results show that maximum system lifetimes calculated in different scenarios using efficient compression technique like LPC is better than without using compression technique.

关键词： Body sensor networks linear predictive coding Sensor systems Energy consumption Compression algorithms Energy dissipation Mathematical model Medical services Biomedical equipment Algorithm design and analysis

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A speaker independent isolated turkish syllable recognition system

A speaker independent isolated turkish syllable recognition ...

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IEEE Signal Processing and Communications Applications (SIU)

作者： N. Ikizler I.H. Cavdar G. Ekim Karadeniz Teknik Üniversitesi Elektronik Mühendisliği Bölümü Trabzon Turkey

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Speech synthesis using two-sided linear prediction parameters

Speech synthesis using two-sided linear prediction parameter...

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International Symposium on Speech, Image Processing and Neural Networks

作者： S.H. Leung H.C. Ng K.F. Wong Department of Electronic Engineering City Polytechnic of Hong Kong Hong Kong China

A two-sided linear prediction (TSLP) model is shown to have high prediction gain over the conventional linear prediction (LPC) model [David and Ramamurthi, 1991], while it requires fewer coefficients in modeling. Unfortunately, speech synthesis cannot use the TSLP model directly because it needs future samples which are not available in the process. Autoregressive spectral matching (ARSM) is proposed to render the TSLP model suitable for speech synthesis. Vector sum excitation method is used to generate the excitation to the new model and its performance is comparable to the standard VSELP.< >

关键词： Speech synthesis predictive models Power system modeling linear predictive coding Autocorrelation Equations Speech processing Image processing Neural networks Cities and towns

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Speaker-independent isolated word recognition for telephone voice using phoneme-like templates

Speaker-independent isolated word recognition for telephone ...

引用

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

作者： T. Nomura R. Nakatsu Electrical Communications Laboratoreis NTT Yokosuka Japan

This paper describes a speaker-independent isolated word recognition algorithm for telephone voice and its recognition performance. The recognition algorithm consists of two processes ; dynamic time warping and statistical word discrimination. In the first process, input speech is compared with each word template using the dynamic time warping technique. Multiple word templates are used to deal with speech variations among speakers, where each word template is represented by a sequence of phoneme-like templates. To attain high recognition ability, a new technique for generating word templates is proposed. In the second process, statistical word discrimination is carried out for word candidates which have relatively low reliability in the first process. Discrimination functions are calculated based on statistics of transition tendencies of speech characteristics between adjacent frames, and the final word decision is made. The system was trained using utterances from 1305 speakers and tested with utterances from 259 speakers. The average recognition rate of 96.5% was obtained for a 16-word Japanese vocabulary set.

关键词： Speech recognition Telephony Speech processing linear predictive coding Power measurement Speech analysis Laboratories Statistics System testing Vocabulary

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Efficient algorithm to compute LSP parameters from 10th-order LPC coefficients

Efficient algorithm to compute LSP parameters from 10th-orde...

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

作者： S. Grassi A. Dufaux M. Ansorge F. Pellandini Institute of Microtechnology University of Neuchâtel Neuchatel Switzerland

Line spectrum pair (LSP) representation of linear predictive coding (LPC) parameters is widely used in speech coding applications. An efficient method for LPC to LSP conversion is Kabal's method. In this method the LSPs are the roots of two polynomials P'/sub p/(x) and Q'/sub p/(x), and are found by a zero crossing search followed by successive bisections and interpolation. The precision of the obtained LSPs is higher than required by most applications, but the number of bisections cannot be decreased without compromising the zero crossing search. In this paper, it is shown that, in the case of 10th-order LPC, five intervals containing each only one zero crossing of P'/sub 10/(x) and one zero crossing of Q'/sub 10/(x) can be calculated, avoiding the zero crossing search. This allows a trade-off between LSP precision and computational complexity resulting in considerable computational saving.

关键词： linear predictive coding Polynomials Interpolation Speech coding Computational complexity Filters Stability Narrowband Chebyshev approximation Frequency

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Pitch Synchronous LPC Analysis for Reduction of coding Distortion caused by Misalignment between Synthesis Filter and its Residual

Pitch Synchronous LPC Analysis for Reduction of Coding Disto...

引用

International Conference on Information, Communications and Signal Processing

作者： H. Farsi Department of Electronic and Electrical Engineering Faculty of Engineering University of Birjand Birjand Iran

Several pre-processing algorithms modify the residual speech signal to facilitate efficient estimation of speech model parameters. This, however, can result in misalignment between the modified residual signal and the time-variant linear prediction (LP) filter used during the synthesis stage. The resulting misalignment may cause audible artifacts particularly at onsets when the frequency response of successive LP filters changes rapidly. We propose a new solution to control the LP filter gain at subframes. This technique is performed before and after time modification of speech and therefore is called preanalysis and post-processing. A pitch smoothing technique is used to illustrate the effect of the proposed technique

关键词： linear predictive coding Speech synthesis Signal synthesis Nonlinear filters Pulse amplifiers Speech coding Pulse measurements Delay Speech analysis Distortion

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