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检索条件"主题词=Linear Predictive Coding"

共 2457 条记录，以下是411-420 订阅

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Connectivity-guided predictive compression of dynamic 3d meshes

Connectivity-guided predictive compression of dynamic 3d mes...

引用

IEEE International Conference on Image Processing (ICIP 2006)

作者： Stefanoski, Nikolce Ostermann, Joern Leibniz Univ Hannover Ist Informationsverarbeitung Appelstr 9A D-30167 Hannover Germany

ISBN: (纸本)9781424404810

We introduce an efficient algorithm for real-time compression of temporally consistent dynamic 3D meshes. The algorithm uses mesh connectivity to determine the order of compression of vertex locations within a frame. Compression is performed in a frame to frame fashion using only the last decoded frame and the partly decoded current frame for prediction. Following the predictive coding paradigm, local temporal and local spatial dependencies between vertex locations are exploited. In this framework we present a novel angle preserving predictor and evaluate its performance against other state of the art predictors. It is shown that the proposed algorithm improves up to 25% upon the current state of the art for compression of temporally consistent dynamic 3D meshes.

关键词： mesh compression dynamic 3D mesh coding animation compression prediction methods linear predictive coding non-linear predictive coding

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Lossless and near-lossless compression for mammographic digital images

Lossless and near-lossless compression for mammographic digi...

引用

IEEE International Conference on Image Processing (ICIP 2006)

作者： Scharcanski, Jacob Univ Fed Rio Grande do Sul Inst Informat BR-91501 Porto Alegre RS Brazil

ISBN: (纸本)9781424404810

In this paper, a method for lossless and near lossless compression of large digital mammograms is proposed. This method is based on a predictive coder that uses integer-to-integer operations, and generates a two-layer embedded bit stream (i.e. near-lossless and refinement layer). The maximum tolerated pixel distortion is guaranteed in the near lossless component. Comparisons with other proposed approaches are based on a database of high-resolution 12 bits/pixel digital mammograms. The simulation results indicate that our lossless compression method can offer average bit rates 39, 5%, 24%, 7% and 2, 5% better than PNG, JPEG 2000, JPEG-lossless, and LOCO, respectively. Besides, for the same images, the proposed method offers near lossless compression at relatively low computational cost, providing an average bit rate of 2, 57 bits/pixel and PSNR of 47,77dB.

关键词： mammography image coding adaptive coding linear predictive coding

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Multi-track codebook in low-rate CELP coding

Multi-track codebook in low-rate CELP coding

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IEEE International Symposium on Industrial Electronics

作者： Guerchi, D. Louzi, A. UAE Univ Coll Informat Technol Jimi 1POB 17555 Al Ain U Arab Emirates Ericsson Canada Montreal PQ H4P2N2 Canada

ISBN: (纸本)9781424404964

In this work, we propose a new technique to model the fixed codebook in low-rate code-excited linear prediction coders. The multi-track codebook presented in this paper reduces the coder bit-rate while maintaining a high speech quality. Unlike the algebraic codebook that is adopted in the 8 kb/s G.729 standard, the multi-track representation uses a pulse density of five pulses per 10 ms speech frame, and permits various pulse magnitudes. Four tracks of disjoint pulse positions sets are constructed and evaluated for each analysis frame. However, only the pulses of the optimal track are quantized and transmitted to the receiver. While the search of the best individual pulse positions is limited to one segment interval of 2 ms, the pulse magnitudes are optimized for the whole analysis frame. The excitation signal in the multi-track codebook is encoded with 2.2 kb/s. Informal test listening and objective test measures have revealed that at the same bit allocation, the multi-track codebook results in significantly higher objective and subjective coded speech quality than the algebraic codebook.

关键词： linear predictive coding speech coding testing coder bit-rate disjoint pulse positions high speech quality informal test listening low-rate CELP coding low-rate code-excited linear prediction coders multitrack codebook objective test pulse density

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Rate-distortion optimization in dynamic mesh compression

Rate-distortion optimization in dynamic mesh compression

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IEEE International Conference on Image Processing (ICIP 2006)

作者： Mueller, K. Smolic, A. Kautzner, M. Wiegand, T. Heinrich Hertz Inst Nachrichtentech Berlin GmbH Fraunhofer Inst Telecommun Image Proc Dept Einsteinufer 37 D-10587 Berlin Germany

ISBN: (纸本)9781424404810

Recent developments in the compression of dynamic meshes or mesh sequences have shown that the statistical dependencies within a mesh sequence can be exploited well by predictive coding approaches. Coders introduced so far use experimentally determined or heuristic thresholds for tuning the algorithms. In video coding rate-distortion (RD) optimization is often used to avoid fixing of thresholds and to select a coding mode. We applied these ideas and present here an RD-optimized mesh coder. It includes different prediction modes as well as an RD cost computation that controls the mode selection across all possible spatial partitions of a mesh to find the clustering structure together with the associated prediction modes. The structure of the RD-optimized D3DMC coder is presented, followed by comparative results with mesh sequences at different resolutions.

关键词： linear predictive coding multidimensional coding 3D compression rate-distortion optimization

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Spectral analysis of Arabic speech signals: Cases of children with normal and impaired hearing

引用

Canadian Journal of Electrical and Computer Engineering 2006年第3期31卷 145-148页

作者： Daoud Boutana Messaoud Benidir Département ďélectronique Faculté des sciences de ľingénieur Université de Jijel Jijel Algeria Laboratoire de signaux Université Paris Sud Gif-sur-Yvette France

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Pattern Recognition and Text-Dependent Recognition for a Mobile Robot

Pattern Recognition and Text-Dependent Recognition for a Mob...

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Midwest Symposium on Circuits and Systems (MWSCAS)

作者： Natalia F. Gonzalez Solano Raul E. Torres Muniz Electrical and Computer Engineering Department University of Puerto Rico Mayaguez Puerto Rico

Pattern Recognition and Audio Processing are important aspects in the control and behavior of mobile robots. Mobile robot's action depends on the recognition of visual and audio stimuli in order to reflect intelligent behavior of the robot. This work presents two recognition systems developed using morphological operations and linear predictive coding (LPC) with Backpropagation Neural Networks (BNN) to process visual and audio data respectively. The objective is to design a person tracking system for a mobile robot with text dependent pitch recognition and a visual pattern recognition mechanism. The BNN will awake the robot from the idle position, while the visual stimulation will be used to track the person given the command.

关键词： Pattern recognition Mobile robots Intelligent robots linear predictive coding Text recognition Robustness Speech recognition Robot sensing systems Neural networks Machine vision

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Approximated Correlation Matrix and Pulse Prediction for Fast Algebraic Code-Excited LPC Speech Coders

Approximated Correlation Matrix and Pulse Prediction for Fas...

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Shu-min Tsai Jar-ferr Yang Institute of Computer and Communication Engineering Department of Electrical Engineering National Cheng Kung University Tainan Taiwan

With reducing computational complexity, an approximated correlation matrix of the vocal impulse response is proposed in algebraic-code-excited linear prediction (ACELP) coders. By exploring statistical characteristics, we only need to calculate a small portion of correlation coefficients before ACELP search procedure. If we further combine a pulse position prediction algorithm, we can reduce the arithmetic complexity in pre-computing autocorrelation matrix and the number of pulse position combinations with imperceptible degradation in speech quality performance. The proposed scheme can be applied to all ACELP coders such as ITU G.729 and G.723.1

关键词： linear predictive coding Computational complexity Speech synthesis Bit rate Mean square error methods Search methods Matrices Prediction algorithms Arithmetic Autocorrelation

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Development of Helicopters Sound Detection Systems

Development of Helicopters Sound Detection Systems

引用

National Conference on Radio Science (NSRC)

作者： Hassan Fahmy Hashem Abbas Amin Alexandria Higher Institute of Engineering and Technology Egypt

The presence of a background noise in the helicopter's sound signals can cause un-acceptance degradation in the detection of the performance method. The recent detection method uses artificial neural networks (ANNs), in combination with parametric spectral representation techniques to detect these targets. The feature encoding techniques based on linear prediction coefficients (LPC) have been applied to obtain spectral estimate of the acoustic signals. This paper investigates preprocessing of signal enhancement prior to feature extraction using different wavelet transform functions

关键词： Helicopters Artificial neural networks Feature extraction Pattern classification Acoustic signal detection linear predictive coding Acoustic noise Noise reduction Acoustical engineering Wavelet transforms

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On HMM Speech Recognition Based on Complex Speech Analysis

On HMM Speech Recognition Based on Complex Speech Analysis

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Annual Conference of Industrial Electronics Society

作者： Tatsuhiko Kinjo Keiichi Funaki Graduate School of Science and Engineering University of Ryukyus Okinawa Japan C N Center University of Ryukyus Okinawa Japan

In speech recognition, LPC cepstrum based on LPC or MFCC based on Mel-frequency filter bank are widely used as a feature extraction that determines the performance. However, these are not being regarded as the best feature extraction. In this paper, we introduce a complex speech analysis for an analytic speech signal to HMM speech recognition. A complex speech analysis can estimate more accurate speech spectrum in low frequencies, as a result, it is expected that the speech analysis can perform well as a feature extractor in speech recognition. The MMSE-based time-varying complex AR speech analysis is adopted and the estimated complex parameters are converted to LPCCs and MFCCs as a feature vector for HTK (HMM tool kit) in order to realize the HMM speech recognition. Through continuous speech recognition experiments with the converted LPCCs and MFCCs, it was found that the complex speech analysis method would not perform well than the real one

关键词： Hidden Markov models Speech recognition Speech analysis Feature extraction linear predictive coding Frequency estimation Cepstrum Mel frequency cepstral coefficient Filter bank Cepstral analysis

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Recovery of LSP Coefficient in VoIP Systems using Evolving Takagi-Sugeno Fuzzy Models

Recovery of LSP Coefficient in VoIP Systems using Evolving T...

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International Workshop on Genetic and Evolving Systems, GEFS

作者： Eric Jones Plamen Angelov Costas Xydeas DSP SW Nokia UK Limited Farnborough Hampshire UK Department of Communication Systems InfoLab2l Lancaster University Lancaster UK

In order to deliver real time, high quality voice services in packet based voice system (e.g. voice over Internet protocol, VoIP) system designers must tackle inherent quality problems related to possible packet loss. To combat the inevitable speech quality deterioration resulting from the loss of transmitted packets of speech information, techniques that provide estimates of the lost information that is needed by the speech recovery process are of considerable interest. Furthermore, in VoIP systems employing linear predictive coding (LPC) based speech coders, a significant percentage of the coded speech information represent the values of LPC coefficients and thus a new approach for estimating missing LPC filter coefficients is presented in this paper. This approach employs a new formulation of LSP recovery system architecture where evolving fuzzy rule-based models and particularly so-called evolving Takagi-Sugeno models are deployed to generate the required estimates of missing LSPs. The proposed missing parameters estimation technique is generic and initial experimental results demonstrate its considerable potential in improving the quality of LPC based decoded speech in VoIP applications

关键词： Takagi-Sugeno model Fuzzy systems linear predictive coding Speech processing Speech coding Real time systems Internet telephony Propagation losses Information filtering Information filters

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