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检索条件"主题词=Linear Predictive coding"

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A method to extract articulatory parameters from the speech signal using neural networks

A method to extract articulatory parameters from the speech ...

引用

International Conference on Digital Signal Processing (DSP)

作者： A. Branco A. Tome A. Teixeira F. Vaz Departamento de Blectrónica e Teleconzunicaçōes da Universidade de Aveiro Campus Universitário Aveiro Portugal INESC Campus Universitário Aveiro Portugal

We present a method that uses artificial neural networks for acoustic to articulatory mapping. An assembly of Kohonen (1982) neural nets is used, in the first stage a network maps cepstral values, each neuron contains a subnet in a second stage that maps the articulatory space. The method allows both the acoustic to articulatory mapping, ensuring smooth varying vocal tract shapes, and the study of the nonuniqueness problem.

关键词： Speech Neural networks Cepstral analysis Neurons Tongue Assembly Lips Shape Signal mapping linear predictive coding

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Speech aid for the deaf based on a representation of the vocal tract: the vowel module

Speech aid for the deaf based on a representation of the voc...

引用

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

作者： P.M.T. de Oliveira M.N. Souza Biomedical Engineering Program COPPE Federal University of Rio de Janeiro Brazil

Presents a new method for speech training of deaf and hard-of-hearing people, using a computer accompanied with an A/D sound card. The idea is to produce a visual feedback of the deaf person's voice, also giving information about what must be done in order to correctly produce the required speech sound. The visual representation attempts to reproduce the configuration of the speaker's vocal tract in real time, during vowel articulation. Since changes in this configuration reflect changes of the spectral envelope of the speech signal, we need to solve the inverse problem: obtaining the vocal tract shape from the speech signal. To solve this problem, signal processing techniques such as linear predictive coding were used. We hope that this representation can be more effective than other representations in use today, such as spectra and spectrograms, due to its closer association with the speech production processes.

关键词： Deafness linear predictive coding Speech processing Feedback Resonance Signal processing Tongue Mouth Data acquisition Biomedical engineering

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A new LPC error criterion for improved pitch tracking

A new LPC error criterion for improved pitch tracking

引用

IEEE Workshop on Speech coding For Telecommunications

作者： M.R. Zad-Issa P. Kabal Electrical Engineering McGill University Montreal QUE Canada McGill University Montreal QC CA

In linear predictive coders the output of the LP analysis filter is used to represent the glottal excitation signal. For high pitched voices during nasal sounds or nasalized vowels, the speech signal takes on a sinusoidal shape. The corresponding residual signal has a very low energy and the pitch pulses are weak or absent, resulting in poor pitch tracking. These segments of speech are also characterized by large frame-to-frame variations of the LP coefficients. In this paper we propose a composite formant prediction error criterion leading to a clear track of residual pulses even for for the sinusoid-like speech, while enhancing the smoothness of the filter parameter evolution.

关键词： linear predictive coding Speech coding predictive models Pulse shaping methods Speech analysis Power system harmonics Signal analysis Speech enhancement Filters Interpolation

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Nonlinear filtering of the LPC residual for noise suppression and speech quality enhancement

Nonlinear filtering of the LPC residual for noise suppressio...

引用

IEEE Workshop on Speech coding For Telecommunications

作者： P. Mermelstein Yasheng Qian INEtS-Télécommunications Université du Quàbec Canada

We study enhancing the quality of coded speech by nonlinear filtering the linear-prediction residual to make it more periodic. By attenuating the low-magnitude samples, more of the residual energy can be concentrated in the regularly spaced major excitation pulses. Wideband additive noise in the speech signal is attenuated because its residual energy is spread in time, rather than being concentrated in narrow temporal regions.

关键词： Filtering linear predictive coding Speech enhancement Speech synthesis Speech analysis Nonlinear filters Additive noise Distortion Signal synthesis Attenuation

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Cost-effective implementation of ITU-T G.723.1 on a DSP chip

Cost-effective implementation of ITU-T G.723.1 on a DSP chip

引用

IEEE International Symposium on Consumer Electronics (ISCE)

作者： Sang-Min Lee Sangil Park Yonngbeom Jang DSP System LSI Division Samsung Electronics Company Limited Yongin si Kyunggi South Korea

ISBN: (纸本)0780343719

This paper presents a full-duplex, real-time implementation of ITU-T G.723.1 speech coder using the SSP1820, Samsung's DSP chip which is based on a 16-bit fixed-point Oak core. Optimization methods are proposed in order to reduce the total cycle time consumed in real-time implementation. The multi-pulse maximum likelihood quantization (MP-MLQ) excitation search block which is the most computation-intensive block in the coder is restructured to reduce the algorithmic redundancy. In addition, efficient filtering methods and memory management are utilized for further optimization. The bit-exact verification with the ITU test vectors and performance evaluation aspects are also discussed in this paper.

关键词： Digital signal processing chips Signal processing algorithms Filters linear predictive coding Digital signal processing Optimization methods Speech synthesis Videoconference Decoding Filtering

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A current-based model for the MOS transistor

A current-based model for the MOS transistor

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： A.I.A. Cunha O.C. Gouveia-Filho M.C. Schneider C. Galup-Montoro Laboratório deInstrumentacçaão Eletroônica (LINSE). Deprtamento de Engenharia Eleétrica Universidade Federal de Santa Catarina Florianopolis Santa Catarina Brazil Laboratoire dePhysique desComposants aà Semiconducteurs (LPCS-INPG) Ecole Nationale Superieure d''Electronique et Radioelectriciteé de Grenoble Grenoble France

This paper presents a physics-based model for the MOS transistor, suitable for circuit design and simulation and valid from weak to strong inversion. Each static or dynamic characteristic is accurately described by a ... 详细信息

关键词： Voltage Circuit synthesis Analog integrated circuits Integrated circuit modeling MOSFET circuits Instruments Electronic mail linear predictive coding Parameter extraction Capacitance

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Development of a multiuser and multimedia game engine based on TCP/IP

Development of a multiuser and multimedia game engine based ...

引用

IEEE Pacific Rim Conference on Communications, Computers and Signal Processing

作者： Jaeyong Park Changhyeon Park Department of Computer Engineering Yeungnam University South Korea

ISBN: (纸本)0780339053

According to the progress of high-speed network and multimedia technologies, related new experimental software is being researched and developed. Multiuser games are a kind of network-based software. The authors describe the implementation of a multiuser and multimedia game engine which enables several users to play the same game simultaneously under a multimedia environment. The implemented game engine provides a script language which can be used to describe the game scenario and facilitates the development of a new game. It also supports the separation of multimedia data necessary for clients from the server, which can reduce the network load and maintain the multimedia data effectively.

关键词： Engines TCPIP Programming profession Computer graphics Computer networks Network servers Personal communication networks Computer languages Hardware linear predictive coding

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Spectral transformation for musical tones via time domain filtering

Spectral transformation for musical tones via time domain fi...

引用

IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

作者： S. Yim Yinong Ding A. McCree DSP Research and Development Center Texas Instruments Inc. Dallas TX USA

We present a spectral transformation technique for musical tones. It can be used to modify the brightness or timbre of a musical tone. We perform spectral transformation by modifying the line spectral frequencies or LPC roots of the original spectral envelope and filtering the tone with a spectral transformation filter in the time domain. One of the application is pitch modification where frequency scaling is used to modify the fundamental frequency of a tone and spectral transformation is used to capture the original brightness and timbre. We have applied this spectral transformation technique to restore the original brightness of frequency scaled musical tones produced from a variety of musical instruments, such as piano, trumpet, violin, flute and bassoon by one octave up or down. In all experiments conducted, the pitch modified tones resemble the original brightness and timbre. This technique allows us to control the timbre of musical tones such as brightness, friction or other modulation by manipulating the spectral parameters.

关键词： Frequency Timbre Brightness Filtering linear predictive coding Spectral shape Acoustic distortion Instruments Filters Friction

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Utterance dependent parametric warping for a talker-independent HMM-based recognizer

Utterance dependent parametric warping for a talker-independ...

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

作者： D.J. Mashao J.E. Adcock LEMS Division of Engineering Brown University Providence RI USA

In an effort to improve the recognition performance of talker-independent speech systems, many adaptive methods have been proposed. The methods generally seek to exploit the higher recognition performance rate of talker-dependent systems and extend it to talker-independent systems. This is achieved by some form of placing talkers into several categories, usually using gender or vocal-tract size. We investigate a similar idea, but categorize each utterance independently. An utterance is processed using several spectral compressions, and the compression with the maximum likelihood is then used to train a better model. For testing, the spectral compression with the maximum likelihood is used to decode the utterance. While the spectral compressions divided the utterances well, this did not translate into significant improvement in performance, and the computational cost increase was significant.

关键词： Hidden Markov models Speech recognition Adaptive systems Testing Maximum likelihood decoding Computational efficiency Speech enhancement Shape Character recognition linear predictive coding

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A lattice low-delay code-excited linear prediction speech coder at 16 kb/s

A lattice low-delay code-excited linear prediction speech co...

引用

IEEE Symposium on Computers and Communications (ISCC)

作者： S.H. El-Ramly R. El-Kouatly M.N. Saleh Electronics and Comm. Eng. Department Faculty of Engineering Ain Shams University Cairo Egypt

This paper presents a lattice low-delay code-excited linear prediction speech coder (LLD-CELP) based on the analysis-by-synthesis configuration. The coder achieves a one-way coding delay of less than 2 ms by making both the LPC predictor and the excitation gain backward-adaptive, and by using a small excitation vector size of four samples. The introduction of a lattice filter as a short term predictor, and a perceptual weighting filter has significant advantages, such as fast tracking of speech signal nonstationarities, simple stability verification, and uniform distribution of the computation load. The introduction of backward adaptive prediction gain imbedded in the excitation codebook achieves a lower complexity with the same quality and with the same bit rate.

关键词： Lattices Filters Speech synthesis Speech analysis Delay Vectors Stability Bit rate Decoding linear predictive coding

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