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检索条件"主题词=Linear predictive coding"

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Towards optimal LPC spectral parameters estimation in standard CELP 4800 b/s speech coder

Towards optimal LPC spectral parameters estimation in standa...

引用

IEEE Digital Signal Processing Workshop

作者： M. Markovic M. Obradovic Institute of Applied Mathematics and Electronics Belgrade Serbia Faculty of Technical Sciences University of Novi Sad Novi Sad Yugoslavia

The influence of nonuniform scalar quantization and the specific interpolation procedure of LSP parameters, as proposed in USA FED STD 1016 CELP 4800 b/s speech coder, to the LPC spectra of speech obtained by using four standard LPC methods: autocorrelation with Hamming window, covariance, modified covariance and lattice, is considered. Comparative experimental analysis is done referring to the results of three different spectral measures related to the RMS LOG spectral measure: likelihood ratio, cosh measure and cepstral distance. The experimental analysis shows that, in the sense of the global LPC spectral representation, the best results are obtained by using the modified covariance method. In this way, the orientation to the autocorrelation method with Hamming window, proposed in USA FED STD 1016 CELP coder, is not justified.

关键词： linear predictive coding Parameter estimation Code standards Cepstral analysis Quantization Autocorrelation Interpolation Lattices Speech coding Filters

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Residual-based speech modification algorithms for text-to-speech synthesis

Residual-based speech modification algorithms for text-to-sp...

引用

International Conference on Spoken Language, ICSLP

作者： M. Edgington A. Lowry BT Laboratories Ipswich UK

The paper presents a set of novel algorithms for the signal modification component of concatenative text-to-speech systems. The algorithms described are based around the LPC analysis/synthesis framework, and achieve prosodic modification by time-domain processing of the LPC residual. The modified residual is then recombined with the all-pole spectral estimate to synthesise the new speech signal. The methods differ in the processing applied to the residual signal. The first method uses a modified version of TD-PSOLA, relying on assumptions of decorrelation and spectral flatness to avoid spectral distortion. The second method uses multiple windowing within each pitch period, enabling a given pitch modification to be realised by shifting several windowed segments by small amounts rather than a large shift of a single window. Again the aim is to reduce phase distortion introduced by the time-shifting process. The third method is based on a smoothly varying resampling of the residual, rather than windowed overlap-add. TD-PSOLA and the residual-based methods were subject a informal listening tests both with pitch and time-scaled natural speech, and also integrated into the signal processing stage of the BT Laureate text-to-speech system.

关键词： Speech synthesis Signal processing algorithms linear predictive coding Signal synthesis Signal processing Algorithm design and analysis Time domain analysis Decorrelation Phase distortion System testing

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Code-excited iterated function prediction

Code-excited iterated function prediction

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Zhicheng Wang Global Media Consulting Sunnyvale CA USA

This paper proposes a new prediction method using iterated function systems (IFS) for vector modeling of an arbitrary discrete sequence. This prediction model uses data itself for representing the future speech signal or discrete sequence and is very different from traditional methods such as LPC model or polynomial fits. On the other hand, the proposed model is different from the traditional IFS because of the backward prediction, fixed-length vector mode, and predictive residual vector usage. The model of iterated function prediction has been developed for a new speech coding scheme. The coding scheme presented is novel and unique and has great potential applications.

关键词： predictive models Vectors Fractals Speech synthesis Image coding Interpolation Prediction methods Polynomials Mathematical model linear predictive coding

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Variable frame rate parameter encoding via adaptive frame selection using dynamic programming

Variable frame rate parameter encoding via adaptive frame se...

引用

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

作者： E.B. George A.V. McCree V.R. Viswanathan DSP Research and Development Center Texas Instruments Inc. Dallas TX USA Personal Systems Laboratory Texas Instruments Inc. Dallas TX USA

This paper presents a block-coded VFR strategy that removes the familiar "anchor frame" constraint in favor of a generalized delayed decision analysis. This approach is referred to as adaptive frame selection using dynamic programming (AFS/DP). The AFS/DP algorithm determines optimal breakpoint locations using a perceptually-based performance measure, accommodates fixed bit-rate operation, and operates with fixed delay and low complexity. The AFS/DP algorithm has been implemented in the context of the MELP vocoder at 2400 bits/sec, and has demonstrated comparable performance to block VFR methods, with lower coder delay.

关键词： Encoding Dynamic programming Delay Speech Vocoders Spectrogram Interpolation Instruments linear predictive coding Laboratories

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Switched prediction and quantization of LSP frequencies

Switched prediction and quantization of LSP frequencies

引用

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

作者： H. Zarrinkoub P. Mermelstein INRS-Telecommun. Quebec Univ. Nuns' Island Que. Canada INRS-Telecommunications Université du Quebéc QUE Canada

We present new results on switched prediction and quantization of the line spectral frequencies applicable to high-quality coding of speech at rates below 8 kb/s. The predictor-quantizer exploits both the temporal inter-frame correlations among the line-spectral frequency vectors of successive frames as well as the spatial within-frame correlations. Best results are obtained by separately training two split vector quantizers, one on predictable frames and one on non-predictable frames. Line-frequency differences are found to yield the lowest quantization error for both the high and low order quantizers for the non-predictable condition, as well as the low order quantizer of the predictable condition. With 10 ms frame separation, 18 bit split VQs, one for each mode, with 1-bit mode indication yield a mean spectral distortion near 1 dB. With 20 ms frame separation, 20 bit quantizers are required.

关键词： Frequency Stability Business Speech coding Nonlinear filters Vector quantization Resonance linear predictive coding Decoding

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Processing disjunctions of temporal constraints

Processing disjunctions of temporal constraints

引用

International Workshop on Temporal Representation and Reasoning (TIME)

作者： E. Schwalb R. Dechter Information and Computer Science Department University of California Irvine USA

Describes new algorithms for processing quantitative temporal constraint satisfaction problems (TCSPs). In contrast to discrete CSPs, enforcing path consistency on TCSPs is exponential due to the fragmentation problem. We present an efficient polynomial constraint propagation algorithm, called "loose-path consistency" (LPC), which is shown to improve the performance of backtrack search algorithms by orders of magnitude. The tradeoffs between the effectiveness and efficiency of LPC are analyzed. We report the presence of a phase transition on this domain and perform an empirical evaluation on problems which lie in the transition region.

关键词： linear predictive coding Polynomials Computer science Databases Quality management Algebra Marine vehicles Performance evaluation

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Low-delay subband CELP coding for wideband speech

Low-delay subband CELP coding for wideband speech

引用

IEEE Region 10 International Conference TENCON

作者： W. Tian W.C. Wong C. Tsao Department of Electrical Engineering National University of Singapore Singapore

In this paper we propose low-delay techniques for coding 7 kHz speech using subband coding together with code-excited linear predictive coding (CELP). We compare the use of separate as well as joint index codebooks. Specifically, the joint index subband CELP (JISBC) algorithm is found to provide good quality with processing delay in the range of 2.375 ms to 3.375 ms at corresponding bitrates of 16 kbit/s to 8 kbits/s.

关键词： Wideband Speech coding Codecs Shape Bit rate linear predictive coding Delay Stochastic processes Teleconferencing Internet telephony

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Exploiting interframe correlation in spectral quantization: a study of different memory VQ schemes

Exploiting interframe correlation in spectral quantization: ...

引用

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

作者： T. Eriksson J. Linden J. Skoglund Department of Information Theory Chalmers University of Technology Goteborg Sweden

This paper addresses the problem of efficient transmission of the LSF parameters in speech coding using vector quantization (VQ). By performing a comparison of several memory VQ methods on the same database, we investigate what gains can be achieved by exploiting interframe correlation. The memory VQ methods studied are finite-state VQ and linear predictive VQ. By combining the memory VQ with a fixed memoryless VQ, called the safety-net, further improvements in performance can be obtained. It is found that memory VQ can improve the performance with 3-5 bits compared to memoryless VQ for error-free transmission. The best method in this study is a safety-net extended predictive VQ. For noisy channels, most memory methods perform worse than memoryless VQ, but the safety-net predictive VQ outperforms memoryless VQ for all tested channel error rates, with 4 bits less.

关键词： Vector quantization Databases Speech coding Performance evaluation linear predictive coding Frequency Information theory Performance gain Testing Error analysis

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Fine structure features for speaker identification

Fine structure features for speaker identification

引用

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

作者： C.R. Jankowski T.F. Quatieri D.A. Reynolds Massachusetts Institute of Technology Cambridge MA US MIT Lincoln Laboratories Lexington MA USA

The performance of speaker identification (SID) systems can be improved by the addition of the rapidly varying "fine structure" features of formant amplitude and/or frequency modulation and multiple excitation pulses. This paper shows how the estimation of such fine structure features can be improved further by obtaining better estimates of formant frequency locations and uncovering various sources of error in the feature extraction systems. Most female telephone speech showed "spurious" formants, due to distortion in the telephone network. Nevertheless, SID performance was greatest with these spurious formants as formant estimates. A new feature has also been identified which can increase SID performance: cepstral coefficients from noise in the estimated excitation waveform. Finally, statistical tools have been developed to explore the relative importance of features used for SID, with the ultimate goal of uncovering the source of the features that provide SID performance improvement.

关键词： Frequency estimation Speech Frequency measurement Telephony Energy measurement Pulse measurements Degradation linear predictive coding Electrostatic precipitators Frequency modulation

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On the use of residual cepstrum in speech recognition

On the use of residual cepstrum in speech recognition

引用

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

作者： Jialong He Li Liu G. Palm Abteilung Neuroinformatik University of Ulm (EBS) Germany

In speech recognition based on LPC analysis the prediction residues are usually ignored, only the LPC-derived cepstral coefficients (LPCC) are used to compose feature vectors. In this study, a number of parameters (called the residual cepstrum or RCEP) were calculated from these residues and their effectiveness for speech recognition was evaluated. It was shown that the RCEP do contain useful information, in particular, they are complementary to the LPCC. In an evaluation experiment, if the LPCC were used jointly with a few RCEP coefficients, the recognition rate of the English E-set letters was improved from 54% to 67% and from 69% to 71% by the use of HMMs based recognizer and the DTW based recognizer, respectively. In addition, Mel-scaled FFT based cepstrum (MFCC) was found to be superior to LPCC.

关键词： Cepstrum Speech recognition linear predictive coding Speech analysis Mel frequency cepstral coefficient Hidden Markov models Cepstral analysis Poles and zeros Helium Parameter estimation

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