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Showing papers on "Cepstrum published in 1982"


Proceedings ArticleDOI
03 May 1982
TL;DR: It is shown in particular that the comb method is computationally more efficient, gives a better fundamental frequency resolution, and can use smaller duration windows for the initial spectral analysis.
Abstract: Most reliable methods of detecting pitch in the speech signal are based on the assumed periodicity found in the voiced speech spectrum (cf. the cepstrum method). Due to recent hardware developments, this approach is becoming more attractive since the prerequired harmonic analysis can now be easily performed in real time. The spectral comb correlation, a new method based on spectral analysis, was recently introduced (5). Rather than performing a second spectral analysis on the log power spectrum, this approach relies on the correlation between the power spectrum and a spectral comb with "teeth" of decreasing amplitude and variable teeth intervals. The maximum of this correlation function is obtained when the width of the teeth intervals corresponds to the periodicity of the spectrum. The spectral comb method is compared to the cepstrum on both theoretical and experimental levels. It is shown in particular that the comb method is computationally more efficient, gives a better fundamental frequency resolution, and can use smaller duration windows for the initial spectral analysis.

72 citations


Proceedings ArticleDOI
01 Jan 1982
TL;DR: Speech quality for voiceband CODECs was evaluated by subjective and objective quality measures and it was concluded that the LPC Cepstrum Distance measure had best correspondence to Mean Opinion Score, among the objective measures studied.
Abstract: This paper describes objective quality measures to evaluate speech quality for various kinds of voiceband CODECs in common. The voiceband CODECs studied were PCM, ADM, ADPCM, ATC (Adaptive Transform Coding) and APC-AB (Adaptive Predictive Coding with Adaptive Bit Allocation). First, several objective quality measures in time and frequency domain were defined. They were SNR, Segmental SNR, Spectral Distortion, LPC Cepstrum Distance, COSH, Likelihood Ratio and Weighted Likelihood Ratio. Second, speech quality for voiceband CODECs were evaluated by subjective and objective quality measures. The subjective measures used were based on opinion test and articulation test. Finally, the relationship between objective measures and subjectively evaluated values was studied. It was concluded that the LPC Cepstrum Distance measure had best correspondence to Mean Opinion Score, among the objective measures studied. It was also concluded that the Wighted Likelihood Ratio measure had best correspondence to Articulation Score.

28 citations


Journal ArticleDOI
TL;DR: A new adaptive algorithm based upon a least square criterion with a weighting factor is presented and shown to be quite useful for estimating ARMA parameters together with input in speech analysis.
Abstract: A new adaptive algorithm based upon a least square criterion with a weighting factor is presented and shown to be quite useful for estimating ARMA parameters together with input in speech analysis. The estimator of both the input pulse train for voiced speech and the input white noise for unvoiced speech are easily obtained from the prediction errors by using this new adaptive algorithm. When these estimated inputs are used as the input of the model to be estimated, the influence of the pitch can be eliminated from the estimated ARMA parameters. By using this method the accuracy of formant and antiformant estimators is shown experimentally in comparison with LPC and cepstrum estimators.

27 citations


Journal ArticleDOI
TL;DR: In this paper, a parametric model of multipath propagation effects on the frequency spectrum is introduced and an optimization technique is used to estimate the coefficients of the model, and the attenuation estimation performance of model estimation technique, the frequency shift technique, and a cepstral liftering technique are compared.
Abstract: Estimates of ultrasonic attenuation based on first‐order or moment spectral estimators are often confused by multipath propagation effects. The magnitude of error in attenuation estimates due to multipath for the frequency shift technique is described. A parametric model of multipath propagation effects on the frequency spectrum is introduced and an optimization technique is used to estimate the coefficients of the model. The attenuation estimation performances of the model estimation technique, the frequency shift technique, and a cepstral liftering technique are compared.

12 citations


Journal ArticleDOI
TL;DR: In this article, existing correlation methods for transit time measurements and limitations for their application are systematically analized and alternative new methods are proposed which are more generally and sometimes easier to apply, conditions for the applicability of the various correlation techniques are specified.

5 citations


Proceedings ArticleDOI
01 May 1982
TL;DR: This work addresses the development of a reliable, high accuracy text-independent speaker recognition system for a small population, with the reference parameters characterizing each speaker obtained from short segments of speech.
Abstract: This work addresses the development of a reliable, high accuracy text-independent speaker recognition system for a small population, with the reference parameters characterizing each speaker obtained from short segments of speech. Initially the potential for speaker discrimination of several different vocal parameter sets was investigated. These included the LPC, Reflection, Cepstrum and Log Area Ratio coefficients, speech power spectrum parameters and the inverse filter spectral coefficients. It was then decided to use any two parameter sets in a composite decision-making scheme. A "repeat feature" was incorporated into the speaker recognition system, whereby a speaker was asked to read a fresh test speech segment if the decisions made by using the two different parameter sets individually were not coincident. Test results indicate that a significant improvement in accuracy is realizable.

4 citations


Proceedings ArticleDOI
07 Dec 1982
TL;DR: In this paper, a new approach to the problem of imaging through a time-varying randomly inhomogeneous medium is presented, which assumes that an ensemble of short exposure images of a single fixed scene or object is available.
Abstract: A new approach to the problem of imaging through a time-varying randomly inhomogeneous medium is presented. This method assumes that an ensemble of short exposure images of a single fixed scene or object is available. It also assumes that knowledge of the average short-term optical transfer function is either known or estimated empirically from a point reference source. Restoration is performed by averaging a modified log gradient of the observed image spectra. This log gradient operator is an exact, mapping of discrete convolution into finite-field addi-tion, and is closely related to the complex "cepstrum," popularized by Oppenheim, It is shown to be invertible to within a constant multiplicative factor by a series of eigenvector computations. The mathematics of this gradient operator are developed, and two-dimensional simulations are presented.

3 citations


Journal ArticleDOI
TL;DR: In this article, an electrooptical power cepstrum analyzer, which displays echo amplitudes and delay times in real-time, is described, and the basic system approach is to use optics to provide the power spectrums and electronics to compute the logarithm function.
Abstract: The power cepstrum (power spectrum of the log of the power spectrum) of a signal in a multipath environment provides information on echo amplitudes and delay times. An electrooptical power cepstrum analyzer, which displays this information in real time, is described. The basic system approach is to use optics to provide the power spectrums and electronics to provide the logarithm function. The design and experimental results of a working system are presented.

2 citations


Proceedings ArticleDOI
01 May 1982
TL;DR: A technique based on fitting splines to the phase derivative curve is presented for the efficient and reliable computation of the two-dimensional complex cepstrum and makes use of several computational strategies within the Tribolet's phase unwrapping algorithm.
Abstract: A technique based on fitting splines to the phase derivative curve is presented for the efficient and reliable computation of the two-dimensional complex cepstrum. The technique is an adaptive numerical integration scheme and makes use of several computational strategies within the Tribolet's phase unwrapping algorithm. An application of the complex cepstrum in testing the stability of two-dimensional recursive digital filters is considered. Susceptibility of the computation of complex cepstrum to slight changes in the coefficients of a two-dimensional array is studied. Several examples of stable and unstable two-dimensional quarter-plane and non-symmetric half-plane recursive digital filters are presented.

2 citations


01 Jan 1982
TL;DR: In this paper, the authors define the requirements for an on-line transmission vibration detection system that could be incorporated on an existing final test stand, which is used to assess transmission noise in order that production tracking of noise levels is possible.
Abstract: This paper is an attempt to define the requirements for an on-line transmission vibration detection system that could be incorporated on an existing final test stand. The purpose of the system is to assess transmission noise in order that production tracking of noise levels is possible. A series of experiments was conducted to define: the best sensor to detect noise levels, the optimum mounting location of the sensor, the ability of a system to discern implanted faults, and the correlation of the vibration levels from an in-plant measurement system to in-vehicle human ratings of transmission noise levels. Results of these tests showed: (1) an accelerometer mounted on the transmission housing offered the best signal clarity in terms of preselected criteria versus a microphone or monitor of fluid pressures, (2) in-plant test stand vibration readings did not correlate well with human ratings of noise when the same units were driven in vehicles, (3) the frequency spectrum plots showed a tendency for the defects to give spectra with higher amplitudes at the gear meshing frequency, (4) signal repeatability on the test stand is excellent, and (5) a technique known as Cepstrum analysis offers promise as a method to reduce the vibration data from the test stand to improve signal discrimination capability. A discussion of a possible production system is presented.

1 citations


Journal ArticleDOI
TL;DR: In this paper, a speech synthesis system using the generalized cepstrum, which is defined as the Fourier coefficients of the generalized logarithmic spectrum, is presented. And the synthesis system in this paper is fully compatible with the one based on the conventional cepstrum method.
Abstract: As a speech analysis method using the generalized cepstrum there is a method based on a generalized logarithmic function instead of a logarithmic function as an evaluation scale This paper presents a speech synthesis system using the generalized cepstrum, which is defined as the Fourier coefficients of the generalized logarithmic spectrum The direct approximated digital filter using the Pade approximation, which is the inverse function of the generalized logarithmic function, is used as a synthesis filter Since this filter not only gives a good approximation of the desired generalized logarithmic spectrum by a generalized cepstrum but also since the values of generalized cepstra correspond to the filter coefficients as they are, it is easy to determine the synthesis filter coefficients Since the synthesis system in this paper is fully compatible with the one based on the conventional cepstrum method, it is possible to synthesize speech by the generalized cepstrum without changing the speech synthesis system based on the conventional cepstrum method Also, using this synthesis system, it is possible to synthesize speech by the conventional cepstrum method as a special case of this method With proper choice of the parameters of the generalized logarithmic function, the synthesized speech by this method is of higher quality than that by the conventional cepstrum method

Proceedings ArticleDOI
01 May 1982
TL;DR: The development of an efficient, general method for the design of 2-D recursive digital filters that effectively uses the 1-D filter theory and is used for stabilizing unstable filters is discussed.
Abstract: This paper discusses the development of an efficient, general method for the design of 2-D recursive digital filters. It effectively uses the 1-D filter theory. The technique employs the McClellan Transformation (MT). Methods are suggested for the design of McClellan Transformation Coefficients (MTC). A stability error criteria is formed on the basis of Complex Cepstrum (CC) and it guarantees the stability of the filter. Group delay and magnitude characteristics are further refined using nonlinear optimization. The method can also be used for stabilizing unstable filters. Numerical implementation and results are presented with an example.

Proceedings ArticleDOI
01 May 1982
TL;DR: The quantization of the system parameters and its effect on the system performance in terms of spectral mismatch are considered and the results are used to develop a cepstral residual vocoder system for 4.8 kbps transmission of speech.
Abstract: The cepstral representation of the LPC residual signal was found in an earlier study to provide a convenient framework for efficiently representing the main spectral components of the residual signal. The analysis-synthesis system developed uses in general poles, zeros and cepstral residual terms to better represent the input speech signal spectrum and achieves improved synthesized speech quality. In this work we consider the quantization of the system parameters and its effect on the system performance in terms of spectral mismatch. The results are used to develop a cepstral residual vocoder system for 4.8 kbps transmission of speech. The system is based on an 8-pole LPC model and the cepstral representation of the residual signal. It applies optimal uniform quantization to each term of the cepstral residual and obtains most of the attainable spectral matching at this useful rate. The study is based on computer simulations with telephone bandwidth speech.

Journal ArticleDOI
TL;DR: In this article, the relation between spectral distortion and the quality of synthesized speech in a speech analysis-synthesis system based on the cepstrum method (cepstral vocoder) is described.
Abstract: The relation between spectral distortion and the quality of synthesized speech in a speech analysis-synthesis system based on the cepstrum method (cepstral vocoder) is described. In this system, the true logarithmic spectral envelope is estimated by an improved cepstral method in the analysis part and a logarithmic amplitude characteristic approximated filter is used in the synthesis part. The transmission rate for spectral information is reduced using the differential of the cepstrum due to the differential of the spectral envelope, because the spectra do not change very rapidly. The preference score by pair comparison tests is employed as a subjective evaluation and spectral distortion is used as an objective evaluation to establish the relations among the quantization width, word length, frame rate, cepstrum order, spectral distortion and synthesized speech quality. Furthermore, the factors of spectral distortion and its characteristics are clarified and it is shown that spectral distortion can be estimated from the transmission condition. The result is that 2.8-kbit/s, high-quality synthesized speech can be obtained by this synthesis system.