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检索条件"主题词=Linear Predictive coding"

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Pitch-synchronous linear-prediction analysis by synthesis with reduced pulse densities

Pitch-synchronous linear-prediction analysis by synthesis wi...

引用

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

作者： D. Guerchi Y. Qian P. Mermelstein INRS-Télécommunications Université du Quàbec Verdun QUE Canada

An important step toward achieving a high-quality 4 kb/s speech codec is reducing the coding-rate of the stochastic codebook component to near 2 kb/s. The increased reconstruction error in the residual that such low-rate quantization implies motivates the search for techniques that reduce the perceptibility of the errors in the reconstructed signal. Pitch-synchronous estimation of the linear-prediction filter and pitch-synchronous updating of the adaptive codebook reduce the coefficient-estimation error and increase the relative contribution of the adaptive codebook component to the synthesized signal, thereby reducing audible noise. However, pitch synchronous analysis normally results in a variable-rate coder. To obtain a fixed-rate representation, we introduce an efficient representation of the stochastic codebook component using a pulse density of one pulse per 2 ms and signed magnitudes specified by 2 bits per pulse-pair. The resulting reconstructions are evaluated for CELP coders corresponding to classical and generalized-pitch-predictor designs. In both cases speech quality comparable to 8 kb/s G.729 is achieved.

关键词： Speech synthesis Signal synthesis linear predictive coding Stochastic processes Frequency estimation Speech codecs Quantization Adaptive filters Estimation error Acoustic noise

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Spectral modification for concatenative speech synthesis

Spectral modification for concatenative speech synthesis

引用

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

作者： J. Wouters M.W. Macon Center for Spoken Language Understanding Oregon Graduate Institute Beaverton OR USA

Concatenative synthesis can produce high-quality speech but is limited to the allophonic variations and voice types that were captured in the database. It would be desirable to modify speech units to remove formant discontinuities and to create new speaking styles, such as hypo- or hyper-articulated speech. Unfortunately, manipulating the spectral structure often leads to degraded speech quality. We investigate two speech modification strategies, one based on inverse filtering and the other on sinusoidal modeling, and we explain their merits and shortcomings for changing the spectral envelope in speech. We then propose a method which uses sinusoidal modeling and represents the complex sinusoidal amplitudes by an all-pole model. The all-pole model approximates the sinusoidal spectrum well, both in the amplitude and in the phase domain. We use the sinusoidal+all-pole model to control the spectral envelope in recorded speech. High-quality modified speech is generated from the model using sinusoidal synthesis. A perceptual test was conducted, which shows that the model was effective at changing vowel identities and was preferable over residual excited LPC.

关键词： Speech synthesis linear predictive coding Filtering Databases Natural languages Filters Frequency Bandwidth Degradation Testing

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Recursive LPC spectrum coding-a classified VQ approach

Recursive LPC spectrum coding-a classified VQ approach

引用

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

作者： F. Norden J. Samuelsson P. Hedelin Inf. Theory Lab. Chalmers Univ. of Technol. Goteborg Sweden Information Theory LabDepartment of Signals and Systems Chalmers University of Technology Gothenburg Sweden Chalmers tekniska hogskola Goteborg SE

We propose a recursive coding scheme for spectral parameters. The coder utilizes a Gaussian mixture model describing the PDF of pairs of source vectors. From this model we can, at each coding instant, calculate a conditional PDF for the vector to be coded given the previously quantized vector. Knowing this conditional PDF, we can use high rate theory to design an optimal codebook. To reduce complexity we employ a classified vector quantizer structure. The mixture components with highest probability around the incoming vector are detected and the corresponding codebooks are searched. A pair of indices, representing the chosen component and codevector, is transmitted. Experiments show a small degradation in performance due to the imposed structural constraint. When compared with a standard predictive vector quantization (PVQ), slightly superior performance, both objectively and subjectively, is observed. Further, close to transparent spectral coding is achieved at 16 bits per frame when employed in a sinusoidal vocoder.

关键词： linear predictive coding Decoding Information theory Degradation Vocoders Speech coding Vector quantization Gaussian processes Spectral analysis Stochastic processes

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Electrocardiogram data compression algorithm based on the linear prediction of the wavelet coefficients

Electrocardiogram data compression algorithm based on the li...

引用

IEEE International Conference on Electronics, Circuits and Systems (ICECS)

作者： M. Abo-Zahhad S.M. Ahmed A. Al-Shrouf Electronics Engineeringineering Department Hijjawi Faculty For Engineering Technology Yarmouk University Irbid Jordan Computer Science Department Faculty of Science Zarka Private University Zarqa Jordan Electrical and Computer Engineeringineering Department Faculty of Engineering Applied Science University Amman Jordan

ISBN: (纸本)0780365429

This paper describes a new algorithm, based on the compression of the linearly predicted residuals of the wavelet coefficients, for electrocardiogram (EGG) compression. The main goal of the algorithm is to reduce the bit rate while keeping the reconstructed signal distortion at a clinically acceptable level. The input signal is divided into blocks and each block goes through a discrete wavelet transform; then the resulting wavelet coefficients are linearly predicted. In this way, a set of uncorrelated transform domain signals is obtained. These signals are compressed using modified run-length and Huffman coding techniques. The error corresponding to the difference between the wavelet coefficients and the predicted coefficients is minimized in order to get the best predictor. The method is assessed through the use of percent residual difference (PRD) and visual inspection measures. By this compression method small PRD with high compression ratio and low implementation complexity are achieved.

关键词： Data compression Electrocardiography Signal processing algorithms Wavelet transforms Wavelet coefficients linear predictive coding Distortion Discrete wavelet transforms Inspection Signal processing

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Multi-layer scalable LPC audio format

Multi-layer scalable LPC audio format

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： A. Jbira A. Kondoz Centre for Communication Systems Research University of Surrey Guildford Surrey UK

This paper presents a scalable audio format, called "multi-layer scalable LPC audio format", that addresses similar functionalities of MPEG-4. The format offers different levels of data rate and audio quality, and answers to the most important requirements of transmission and storage purposes, such as channel error robustness, cell loss robustness, low delay, and playback control. It operates in four modes. The first mode is based on a modified version of the LD-CELP algorithm, in which each 6 samples are represented by one single byte. In order to improve the signal-to-noise ratio (SNR), additional enhancement layers are embedded in the bit stream to allow higher quality at higher bit rates. The resultant bit rates are integer-multiple of 10.67 kbps. The other three modes use QMF splitting to two, four and eight subbands. These modes allow efficient representation of wideband audio and speech signals, and offer extension layers of 5.33 and 2.66 kbps. A simple and efficient header structure is embedded in the bitstream to allow the decoding process even in channel error conditions and even when the bitstream has been down-scaled somewhere during the transmission but has not been acknowledged to the decoder. Comparison results are conducted with respect to MPEG and ITU standards.

关键词： linear predictive coding Robust control Bit rate Decoding MPEG 4 Standard Error correction Propagation losses Delay Signal to noise ratio Streaming media

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An overview of text-to-speech synthesis

An overview of text-to-speech synthesis

引用

IEEE Workshop on Speech coding Proceedings

作者： A. Acero Speech Technology Group Microsoft Corporation Redmond WA USA

Summary form only given. The article gives an overview of text-to-speech (TTS) technology and a description of some issues of potential interest to speech coding experts. After motivation for the use of TTS technology, it describes the general architecture of a text-to-speech system with particular emphasis on the speech synthesis component. Both formant synthesis and concatenative synthesis are presented, offering different degrees of flexibility and quality. Several well-known speech coding techniques (including LPC vocoders, waveform interpolation, harmonic coding, and layered coding) have been used in speech synthesis. It explains how they have been applied, and the advantages and limitations of those techniques when used in speech synthesis. The main goal is to increase cooperation between the speech coding community and the TTS community, and in particular to motivate the need for speech coding algorithms that meet the requirements of the next generation speech synthesis technology.

关键词： Speech synthesis Speech coding linear predictive coding Vocoders Interpolation

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A novel variable-rate MELP speech coder

A novel variable-rate MELP speech coder

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： Tie Manxia Wang Dusheng Fan Changxin Information Science Institute Xidian University Xi'an China

In MELP coder, LPC coefficients are quantized to 25 bit/frame by MSVQ. The overall bit of MELP is 54 bit/frame, so more than 45% of the required bandwidth is spent on transmission of LPC coefficients. In this paper, based on LPC adaptive forward-backward quantization, a novel variable-rate MELP coder is proposed, in which linear prediction is done by using either the currently (forward LPC) or previously decoded (backward LPC) speech frame. The backward LPC scheme shall be applied, i.e., the LPC coefficients based on the previously decoded optimal speech frame are used to encode the current frame and only the time delay shall be transmitted to the decoder, so, average LPC bit number becomes smaller. Computer simulation shows significant average overall bit rate reduction is achieved without compromising the decoded speech quality.

关键词： linear predictive coding Decoding Bit rate Speech coding Vocoders Quantization Distortion measurement Bandwidth Information science Delay effects

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Improving the performance of a small microphone array at low frequencies using critical band and LPC codebooks

Improving the performance of a small microphone array at low...

引用

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

作者： Yuchang Cao S. Sridharan Speech Research Laboratory RCSAVT School of EESE Queensland University of Technology Brisbane QLD Australia

This paper presents a novel method for improving the low frequency performance of a microphone array by use of a twin codebook consisting of critical band energy and LPC distribution of speech. A modified Wiener filter created using these codebooks eliminates residual noise at low frequencies, which may result from the limitation of the array size or the beamforming algorithm. The preliminary simulation tests have shown that the proposed system is effective, and may provide good performance for those microphone array systems which are installed on a computer in an office, or in a moving vehicle.

关键词： Microphone arrays Frequency linear predictive coding Speech Wiener filter Low-frequency noise Array signal processing Computational modeling Computer simulation System testing

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An 8 Kb/s low-delay CELP speech coder

An 8 Kb/s low-delay CELP speech coder

引用

IEEE International Conference on Electronics, Circuits and Systems (ICECS)

作者： Hongmao Sun Chunyan Wang W.O. Ahmad M.N.S. Swamy Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada Concordia University Montreal QC CA

ISBN: (纸本)0780365429

This paper presents an efficient low-delay CELP speech coder based on a structure given by Chen et el. (1992). The proposed coder can operate at a rate of 8 Kb/s and has an arithmetic complexity that is 20% lower than that of the CELP of Chen et al. with an acceptable increase in the delay. The proposed coder has been tested to provide a good-quality speech.

关键词： Speech Decoding linear predictive coding Bit rate Delay effects Filters Vectors Sun Arithmetic Testing

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Speech recognition using neural networks

Speech recognition using neural networks

引用

IEEE International Conference on Industrial Technology (ICIT)

作者： S.U. Khan G. Sharma P.R.K. Rao Department of Electrical Engineering Indian Institute of Technology Kanpur India Secunderabad India

The paper presents a continuous speech recognition system based on a neural network concept. An articulatory-phonetic feature extraction network (APFEN) is used for extracting articulatory-phonetic features. This is followed by a coarticulation network for reducing the effect of coarticulation present in the continuous speech. Algorithms for segmentation and then identification of phonemes are given.

关键词： Speech recognition Neural networks Feature extraction Paper technology Cepstrum linear predictive coding Humans Reflection Mel frequency cepstral coefficient Frequency domain analysis

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