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检索条件"主题词=linear predictive coding"

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Speech Recognition Using Neural Networks

Speech Recognition Using Neural Networks

引用

International Conference on Signal Processing Systems, ICSPS

作者： T. Lalith Kumar T. Kishore Kumar K. Soundar Rajan SVIST Kadapa India National Institute of Technology Warangal India JNTU Anantapur India

ISBN: (纸本)9781424442133

Speech processing has been an active area for several decades with a wide variety of applications ranging from communications to automatic reading machines. There are many speech recognition techniques, which are based on statistical techniques as well as neural networks. The present work investigates the feasibility of two approaches for solving the problem using neural networks.

关键词： Speech recognition Neural networks Iron Recurrent neural networks Feature extraction linear predictive coding Cepstrum Pattern recognition Sequences Signal processing

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A hybrid input/output spectrum adaptation scheme for LD-CELP coding of speech

A hybrid input/output spectrum adaptation scheme for LD-CELP...

引用

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

作者： Siu-Pun Chui Cheung-Fat Chan Department of Electronic Engineering City University of Hong Kong Hong Kong China

ISBN: (纸本)0780331923

This paper describes a hybrid input/output block adaptation scheme for the quantization of spectral parameters for low-delay CELP coders. In this scheme, line spectrum pair (LSP) coefficients are used as parameters for spectral adaptation and they are partitioned into two segments; LSPs in the first segment which corresponding to low frequency regions are backward adapted from the reproduction speech while LSPs in the second segment are adapted from input speech and quantized as side information to the decoder. The rationale behind this scheme is to compensate for the spectrum distortion introduced by conventional backward adaptation schemes operating at low bit rates. Simulation results show that the hybrid LSP adaptation scheme achieves transparent quantization at 12 bits for each analysis frame of 5 ms.

关键词： Speech coding Delay Quantization Bit rate Speech analysis Acoustic noise Speech processing Speech enhancement linear predictive coding Frequency

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Interframe differential vector coding of line spectrum frequencies

Interframe differential vector coding of line spectrum frequ...

引用

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

作者： E. Erzin A.E. Cetin Department of Electrical and Electronics Engineering Bilkent University Ankara Turkey

In many vocoders LSFs (line spectrum frequencies) are used to encode the linear predictive coding (LPC) parameters. An interframe differential coding scheme is presented for LSFs. The LSFs of the current speech frame are predicted by using both the LSFs of the previous frame and some of the LSFs of the current frame. Then the difference vector resulting from prediction is vector quantized. The proposed scheme is computationally efficient and easy to implement, and can be used in low-bit-rate vocoders.< >

关键词： Frequency linear predictive coding Speech coding Vectors Nonlinear filters Vocoders Speech synthesis Polynomials predictive models Signal synthesis

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Code-excited linear prediction(CELP): High-quality speech at very low bit rates

Code-excited linear prediction(CELP): High-quality speech at...

引用

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

作者： M. Schroeder B. Atal Drittes Physikalisches Institut University of Goettingen Germany AT and T Bell Laboratories Inc. Murray Hill NJ USA

We describe in this paper a code-excited linear predictive coder in which the optimum innovation sequence is selected from a code book of stored sequences to optimize a given fidelity criterion. Each sample of the innovation sequence is filtered sequentially through two time-varying linear recursive filters, one with a long-delay (related to pitch period) predictor in the feedback loop and the other with a short-delay predictor (related to spectral envelope) in the feedback loop. We code speech, sampled at 8 kHz, in blocks of 5-msec duration. Each block consisting of 40 samples is produced from one of 1024 possible innovation sequences. The bit rate for the innovation sequence is thus 1/4 bit per sample. We compare in this paper several different random and deterministic code books for their effectiveness in providing the optimum innovation sequence in each block. Our results indicate that a random code book has a slight speech quality advantage at low bit rates. Examples of speech produced by the above method will be played at the conference.

关键词： Bit rate Technological innovation Speech coding Books Feedback loop Nonlinear filters Speech synthesis predictive models linear predictive coding predictive maintenance

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Least squares-adapted edge-look-ahead prediction with run-length encodings for lossless compression of images

Least squares-adapted edge-look-ahead prediction with run-le...

引用

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

作者： Lih-Jen Kau Yuan-Pei Lin Department Elec. and Control Engineering National Chiao Tung University Hsinchu Taiwan

Many coding methods are more efficient with certain types of images than others. In particular, run-length coding is very useful for coding areas of little changes. Adaptive predictive coding achieves high coding efficiency for fast changing areas like edges. In this paper, we propose a switching coding scheme that will combine the advantages of both run-length and adaptive linear predictive coding (RALP) for lossless compression of images. For pixels in slowly varying areas, run-length coding is used; otherwise LS (least square)-adapted predictive coding is used. Instead of performing LS adaptation in a pixel-by-pixel manner, we adapt the predictor coefficients only when an edge is detected so that the computational complexity can be significantly reduced. For this, we propose an edge detector using only causal pixels. This way, the predictor can look ahead if the coding pixel is around an edge and initiate the LS adaptation in advance to prevent the occurrence of a large prediction error. With the proposed switching structure, very good prediction results can be obtained in both slowly varying areas and pixels around boundaries as we will see in the experiments.

关键词： Image coding Image edge detection Computational complexity predictive coding Encoding linear predictive coding Detectors Bit rate Biomedical imaging Remote sensing

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Network echo canceller based on the practical adaptive filter

Network echo canceller based on the practical adaptive filte...

引用

IEEE International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)

作者： Jae Ha Yoo Dong Yon Kim Department of Electronic Engineering Hankyong National University South Korea

In this paper, a new network echo canceller based on the practical adaptive filter is proposed. The proposed adaptive filter practically modifies the lattice transversal joint (LTJ) adaptive filter. It takes advantage of information in the speech decoder and coefficients of the transversal filter part in the LTJ adaptive filter are updated every other sample instead of every sample. Total complexity of the proposed adaptive filter is lower than that of the transversal filter. And the residual echo signal is decreased by residual echo cancellation using the lattice predictor whose order is less than 10. Computational complexity of the proposed echo canceller is lower than that of the transversal filter but the convergence speed is faster than that of the transversal filter. The performance of the proposed network echo canceller was verified by the experiments using the real speech signal.

关键词： Echo cancellers Adaptive filters Transversal filters Lattices Computational complexity Convergence linear predictive coding Decoding Speech processing Natural languages

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Low bit-rate speech coder based on a long-term model

Low bit-rate speech coder based on a long-term model

引用

IEEE Convention of the Electrical and ELectronic Engineers in Israel

作者： O. Lev D. Malah Dept. of Electr. Eng. Technion-Israel Inst. of Technol. Haifa Israel Department of Electrical Engineering Technion Israel Institute of Technology Israel

We present a low bit rate speech coder based on a long-term model (LTM) for voiced speech, and on the WI coder. In the LTM, a periodic input signal undergoes a time-varying spectral shaping representing the evolution of the pitch-cycle waveform. The resulting signal, which has a fixed pitch period but a time-varying pitch-cycle waveform, is multiplied by a time-varying gain function that represents the variation in speech loudness. The resulting signal then undergoes a time-axis warping, which represents the evolution of the pitch period, yielding the output speech signal. The spectral shaping in the proposed coder is based on WI. In WI, speech (or LPC residual) is observed as a continuously evolving sequence of pitch cycle waveforms. A subset of these waveforms is extracted and coded. In the decoder, after inverse quantization, missing waveforms are synthesized by interpolation. The extracted waveforms are normalized to a fixed length and sequentially aligned using a cyclical shift. Then, a two-dimensional surface, called prototype waveform surface or characteristic waveform (CW) is produced from these waveforms.

关键词： Speech analysis Speech synthesis Surface waves Interpolation linear predictive coding Decoding Energy states Gaussian noise Signal analysis Bit rate

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A new approach of LPC analysis based on the normalization of vocal-tract length

A new approach of LPC analysis based on the normalization of...

引用

International Conference on Pattern Recognition

作者： Huang Ze-Zhen Yan Xiag-Jun Department of radio electronics Tsinghua University Beijing China Dept. of Radio Electron. Tsinghua Univ. Beijing China

ISBN: (纸本)0818608781

An approach to linear prediction coefficient (LPC) analysis based on the normalization of vocal-tract length is presented. The approach is of significance for speech recognition of arbitrary speakers. In this approach, the ratio of two vocal-tract lengths corresponding to a new speaker and a reference one is first estimated from the training speech data of several typical vowels. The LPC parameters normalized on this ratio can then be calculated for any speech data. Compared with previous methods of speech parameter normalization, this approach does not need to estimate formant frequencies and is simple and reliable in theory. Limited experiments on the recognition of nine Chinese vowels for four speakers to indicate that this new approach can achieve 5% to 20% improvements of correct recognition rate.< >

关键词： linear predictive coding Autocorrelation Shape Frequency estimation Fourier transforms Speech Testing Zirconium Maximum likelihood estimation Sampling methods

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TD-PSOLA versus harmonic plus noise model in diphone based speech synthesis

TD-PSOLA versus harmonic plus noise model in diphone based s...

引用

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

作者： A. Syrdal Y. Stylianou L. Garrison A. Conkie J. Schroeter AT and T Research Laboratories Florham Park NJ USA AT and T Bell Laboratories Inc. Holmdel NJ USA

In an effort to select a speech representation for our next generation concatenative text-to-speech synthesizer, the use of two candidates is investigated; TD-PSOLA and the harmonic plus noise model, HNM. A formal listening test has been conducted and the two candidates have been rated regarding intelligibility, naturalness and pleasantness. Ability for database compression and computational load is also discussed. The results show that HNM consistently outperforms TD-PSOLA in all the above features except for computational load. HNM allows for high-quality speech synthesis without smoothing problems at the segmental boundaries and without buzziness or other oddities observed with TD-PSOLA.

关键词： Speech synthesis Testing Speech enhancement linear predictive coding Speech analysis Acoustic noise Synthesizers Spatial databases Smoothing methods Man machine systems

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Line spectrum pair (LSP) and speech data compression

Line spectrum pair (LSP) and speech data compression

引用

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

作者： F. Soong B. Juang Acoustic Research Department Bell Laboratories Murray Hill NJ USA

Line Spectrum Pair (LSP) was first introduced by Itakura [1,2] as an alternative LPC spectral representations. It was found that this new representation has such interesting properties as (1) all zeros of LSP polynomials are on the unit circle, (2) the corresponding zeros of the symmetric and anti-symmetric LSP polynomials are interlaced, and (3) the reconstructed LPC all-pole filter preserves its minimum phase property if (1) and (2) are kept intact through a quantization procedure. In this paper we prove all these properties via a "phase function." The statistical characteristics of LSP frequencies are investigated by analyzing a speech data base. In addition, we derive an expression for spectral sensitivity with respect to single LSP frequency deviation such that some insight on their quantization effects can be obtained. Results on multi-pulse LPC using LSP for spectral information compression are finally presented.

关键词： Data compression linear predictive coding Polynomials Quantization Frequency Speech analysis Dynamic range Information filtering Information filters Stability

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