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检索条件"主题词=linear predictive coding"

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Improving LPC analysis of noisy speech by autocorrelation subtraction method

Improving LPC analysis of noisy speech by autocorrelation su...

引用

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

作者： C. Un K. Choi Department of Electrical Science Korea Advanced Institute of Science and Technology Seoul South Korea Central Research Laboratory Gold Star Company Limited Seoul South Korea

A robust linear predictive coding (LPC) method that can be used in noisy as well as quiet environment has been studied. In this method, noise autocorrelation coefficients are first obtained and updated during non-speech periods. Then, the effect of additive noise in the input speech is removed by subtracting values of the noise autocorrelation coefficients from those of autocorrelation coefficients of corrupted speech in the course of computation of linear prediction coefficients. When signal-to-noise ratio of the input speech ranges from 0 to 10 dB, a performance improvement of about 5 dB can be gained by using this method. The proposed method is computationally very efficient and requires a small storage area.

关键词： Speech analysis linear predictive coding Autocorrelation Speech enhancement Noise cancellation Additive noise Testing Working environment noise Degradation Wiener filter

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Inter-speaker comparison of LPC acoustic space using a minimax distortion measure

Inter-speaker comparison of LPC acoustic space using a minim...

引用

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

作者： E. Dorsey J. Bernstein Telesensory Speech Systems Telesensory Systems Inc. Palo Alto CA USA

A method is described that measures the range of frequency spectra produced when a person speaks. Our "minimax" method selects a certain size subset from a set of LPC vectors such that the maximum RMS log spectral distance from any vector in the superset to its nearest neighbor in the subset is minimized. The minimum inter-vector distance in the subset is, in general, monotonically related to the volume of the region that contains the superset. Passages of English, Japanese, and Spanish were recorded, and analyzed using this method. Results indicate that the range of spectra characteristic of different individual's speech are significantly different. There is less difference, if any, between languages in their measured acoustic ranges. The results are discussed with reference to issues in phonology and speech coding.

关键词： linear predictive coding Extraterrestrial measurements Minimax techniques Acoustic distortion Acoustic measurements Distortion measurement Frequency measurement Nearest neighbor searches Speech Natural languages

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Isolated word recognition using a two-pass pattern recognition approach

Isolated word recognition using a two-pass pattern recogniti...

引用

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

作者： L. Rabiner J. Wilpon Bell Laboratories Murray Hill NJ USA

One of the major drawbacks of the standard pattern recognition approach to isolated word recognition is that poor performance is generally achieved for word vocabularies with acoustically similar words. This poor performance is related to the pattern similarity (distance) algorithms that are generally used in which a global distance between the test pattern and each reference pattern is computed. Since acoustically similar words are, by definition, globally similar, it is difficult to reliably discriminate such words, and a high error rate is obtained. By modifying the pattern similarity algorithm so that the recognition decision is made in two passes, improvements in discriminability among similar words can be achieved. In particular, on the first pass the recognizer provides a set of global distance scores which are used to decide a class (or a set of possible classes) in which the spoken word is estimated to belong. On the second pass a locally weighted distance is used to provide optimal separation among words in the chosen class (or classes) and the recognition decision is made on the basis of these local distance scores. For a highly complex vocabulary (letters of the alphabet, digits, and 3 command words) recognition improvements of from 3 to 7 percent were obtained using the two-pass recognition strategy.

关键词： Pattern recognition Vocabulary Speech linear predictive coding Acoustic testing

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Automatic generation of mixed excitation in a linear predictive speech synthesizer

Automatic generation of mixed excitation in a linear predict...

引用

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

作者： S. Holm Division of Telecoamninications Norwegian Institute of Technology Trondheim Norway

The main objective of this work has been to add the model for fricative excitation to the LPC synthesis model. From the LPC model one finds the acoustic tube section with the greatest constriction and adds a modulated noise signal. The results from this model demonstrate that one is able to produce noise bursts at the right time instants that are shorter than the frame length. This gives a more natural sound for certain phonemes, but adds a quantization type of background noise.

关键词： Speech synthesis Synthesizers Filters linear predictive coding Signal synthesis predictive models Acoustic noise Transfer functions Quantization Background noise

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The implementation of a speech synthesis algorithm

The implementation of a speech synthesis algorithm

引用

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

作者： G. Amir R. Gregorian G. Edwards American Microsystems Inc. Santa Clara CA USA

The application of integrated circuit technology to Speech Synthesis and Recognition represents an important development in the field. This paper describes a complete Speech Synthesis System on a single chip. The considerations involved in the choice of a compatible algorithm, machine word length and coefficient accuracy are discussed. The device contains a 32 word vocabulary and an innovative implementation of the LPC lattice structure. It operates at a variable bit rate to provide high quality speech with low bit storage requirements. The software supporting the Speech Synthesis System is also described.

关键词： Speech synthesis Speech processing Large scale integration Speech analysis linear predictive coding Vocabulary Bit rate Synthesizers Integrated circuit synthesis Semiconductor device manufacture

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An LPC vocoder for efficient implementation

An LPC vocoder for efficient implementation

引用

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

作者： F. Sugiyama M. Nakamura Toshiba Research and Development Center Kawasaki Japan

linear prediction has been widely used in speech coding. However, a large amount of calculation is necessary for parameter extraction by the autocorrelation method, so it is not suited for hardware implementation. This paper proposes a new analysis method, involving iterative method in the lattice circuit for successive K parameter approximations. In the iterative method, the improved hybrid algorithm was found best, from the viewpoint of high speed convergence for speech. For pitch extraction, partly divided average magnitude difference function (AMDF) method was used. Using these two methods, a vocoder was developed which requires only a small amount of calculation and is suited for hardware implementation (e.g. LSI). Good speech quality was obtained.

关键词： linear predictive coding Vocoders Hardware Iterative methods Speech coding Parameter extraction Autocorrelation Lattices Circuits Iterative algorithms

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Comparison of several clustering algorithms for data rate compression of LPC parameters

Comparison of several clustering algorithms for data rate co...

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

作者： J. Marr Electrical Engineering Department University of Alabama Huntsville Huntsville AL USA

The purpose of this research was to investigate the effectiveness of several clustering algorithms for separation of LPC speech segments. Four thousand frames of speech parameters were divided into thirty clusters using four clustering algorithms and three initial seed point selection methods. A second-order and an eithth-order norm in the area function parametric domain were used for distance measures. For single-pass clustering algorithms, using every 130th frame as a seed point resulted in a substantially lower error than that given by using the first thirty frames as seed points. The benefit was noticable for using a more complex algorithm for generating the initial seed points. For iterative clustering algorithms, the initial allocation had negligible effect on the final error and on the number of iterations.

关键词： Clustering algorithms linear predictive coding Iterative algorithms Area measurement Speech recognition Vocoders Humans Clustering methods Testing Time measurement

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Application of the PDSP chip set to LPC synthesis

Application of the PDSP chip set to LPC synthesis

引用

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

作者： E. Dorsey J. Caldwell Telesensory Speech Systems Division Telesensory Systems Inc. Palo Alto CA USA

Due to its programmability, the Programmable Digital Signal Processor (PDSP)chip set lends itself to the implementation of a variety of speech synthesis functions. When programmed as an all-pole lattice filter used in LPC synthesis, the PDSP allows the user to exploit quality-enhancement techniques such as mixed sourcing, pitch- or time-synchronous updating, variable frame and filter length, interpolation, various quantizing schemes, and adjustable speaking rate. Since the quantizing algorithm is under user control, the encoded bit rate may range from 1100 to 28,000 bits per second according to speech quality requirements. The paper discusses these techniques and a commercially available LPC speech synthesis circuit board which illustrates typical PDSP application.

关键词： linear predictive coding Speech synthesis Signal synthesis Filters Digital signal processors Lattices Interpolation Signal processing algorithms Bit rate Printed circuits

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A model for short-time phase prediction of speech

A model for short-time phase prediction of speech

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

作者： L. Almeida J. Tribolet Centro de Anàlise e Processamento Sinais Instituto Superior Técnico Lisboa Portugal

Prediction plays a key role in many signal processing applications. linear Prediction has, in particular, been extremely useful to the development of digital speech processing techniques and applications. There is however a growing need for improved forms of prediction. We discuss, in this paper, a form of non-linear prediction, namely, the prediction of the phase of speech signals. This study is conducted within a short-time analysis/synthesis framework and is based upon a new treatment of the classical speech production model. Experimental data are presented confirming the theoretical results. Finally the use of phase prediction to low-bit rate, high-quality coding applications is discussed.

关键词： predictive models Nonlinear filters Speech analysis linear predictive coding Speech processing Speech synthesis Frequency Signal processing Signal synthesis Speech recognition

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Recent developments in vector quantization for speech processing

Recent developments in vector quantization for speech proces...

引用

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

作者： D. Wong B. Juang A. Gray Signal Technology Inc. Santa Barbara CA USA

Vector Quantization is applied to modify a 2400 bps LPC vocoder to operate at 800 bps, while retaining acceptable intelligibility and naturalness of quality. The design of this speech compression system is discussed and compared to other very low bit rate vocoders. Advantages of vector quantization over a scalar technique are examined in detail, and several new properties are presented.

关键词： Vector quantization Speech processing linear predictive coding Filters Bit rate Vocoders Distortion measurement Training data Background noise Rate-distortion

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