Multipulse Excitation Research Articles

5283525 Method and device for the volume-selective extraction of a magnetic resonance spectrum by Hartmann-Hahn transfer

The quantitative assessment of lactic acid in tissue is an important goal for in vivo volume-selective NMR spectroscopy to aid in the noninvasive diagnosis of oxygen deficiency or other metabolic disorders. PRESS localized 1H spectra provide comparatively high signal-to-noise ratio from small volume elements in a single acquisition mode. The quantification of lactate after multipulse excitation is not trivial due to the J-coupling characteristics which do not occur for the substances serving as references. The influence of the timing scheme and of the quality of the refocusing pulses was systematically evaluated for the lactate resonances by volume-selective measurements. Gaussian pulses, Hanning-filtered sinc pulses, and numerically optimized RE-BURP-pulses were applied for refocusing the magnetization in the PRESS sequence and the effects on the lactate AX3 spin system were compared. For these pulses, sequence parameters are presented providing high sensitivity to lactate signals. Timing schemes are shown which provide good quantification of lactate, even in cases with B1-inhomogeneities or slight misadjustment of the transmitter amplitude. The combination of both echo times in the double-echo sequences clearly influences the signal characteristics of lactate at overall echo times near TE = 145 and 290 ms, which may result in pure in-phase magnetization for this weakly coupled homonuclear system. Numerically optimized refocusing pulses (RE-BURP) provided up to 50% higher signal ratio of the methyl protons of lactate to uncoupled nuclei than the often used Hanning-filtered sinc pulses.

A variable rate multipulse speech coder for CDMA cellular systems

Variable Rate (VR) speech coders are classified into: source-controlled VR coders where the rate is selected depending on the local character of the speech, and network-controlled VR coders where an external control signal selects the coding rate. The first category benefits from the variable rate channels used by Code Division Multiple Access (CDMA) mobile communications. The second category is indispensable for the right behaviour of the CDMA systems under conditions as high traffic levels. The VR speech coder presented in this communication exhibits both types of control. The source control is achieved by means of a Voice Activity Detector (VAD) and a phonetic classifier. The network control acts on the selection procedure of the multipulse excitation sequence to the synthesis filter. This is the main advantage of our VR MultiPulse speech coder because by means of an external signal the bit rate can be changed only every 4 msec, without transitions or distortions. Considering one-way communication, six different operating rates can be externally selected ranging from 4.8 to 9.1 kbps for the active frames; an average bit rate of 380 bps is required for the noise frames.

Improved multipulse algorithm for speech coding by means of adaptive boltzmann annealing

AbstractIn this paper we will describe an algorithm, based on the Simulated Annealing technique, for solving the problem of multipulse excitation computation for speech coding applications. Multipulse excitation modeling is described by two set of parameters: pulse locations and amplitudes. The search for optimal pulse locations is the most burdensome task, since there is no closed form solution. Therefore, suboptimal approaches are used in order to maintain a good effectiveness in the parameter estimation while keeping the computational load within reasonable bounds. As compared to the one described in [2], the proposed algorithm achieves an effective improvement in objective performances of more than 1.5 dB. The price that must be paid for this performance improvement is the higher computational cost that this algorithm requires. The objective results of the proposed algorithm will be compared with the results of the algorithm described in [2], and the computational complexity will be analyzed.

Means for improving the speech quality in multi‐pulse excited linear predictive coding

A technique that reconciles the differences between the estimator and the filter of a multi-pulse linear predictive voice encoder achieves a higher quality in the output speech. The technique simultaneously solves for the pulse amplitudes and pitch tap gain to minimize the estimator bias in the multi-pulse excitation and thereby improves, performance of the system. The increased signal-to-noise ratio is accomplished by first modifying the pitch predictor such that the pitch synthesis filter accurately reflects the estimation procedure used to find the pitch tap gain and, second, improving the excitation analysis technique such that the pitch predictor tap gain and pulse amplitudes are solved for simultaneously, rather than sequentially. Neither of these modifications results in an increased transmission rate and they do not significantly increase the complexity of the multi-pulse coding algorithm.

Speech coding using sparse vector codebook and cyclic shift techniques

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Communication system capable of improving a speech quality by a pair of pulse producing units

A second approximation of the multipulse excitation signal is derived from a difference signal developed from use of a first approximation of the multipulse excitation signal. Also, spectrum parameters are weighted by a periodicity measure.

An approach to parameter reoptimization in multipulse-based coders

An algorithm for LPC parameter optimization in multipulse (MP)-linear-prediction-coding (LPC)-based speech coders is presented. It is shown that by taking the nature of the MP excitation signal into account in the LPC parameter computation, it is possible to improve the effectiveness of the LPC model. This results in a better quality of the reconstructed signal in terms of both objective and subjective criteria. The implementation details of the algorithm are discussed, and experimental results are presented. In particular, a comparison with standard MP LPC techniques is given.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Generation of transient inversion gratings with an ultrashort period in media exhibiting a phase memory under multipulse interaction conditions

Multipulse excitation of a resonant medium by a laser field is considered. It is shown that one stable long-lived grating with an ultrashort period, i.e., a period much shorter than the wavelength of the exciting laser radiation, can form in such a medium. A study is reported of the interaction of light with gratings of this type and of generation of transient signals by these gratings.

Method of encoding speech signals using a multipulse excitation signal having amplitude-corrected pulses

In a digital speech encoding system wherein the transmitter substitutes for the original speech signal to be encoded (1) certain parameters supplied by an analysis circuit defining, within successive time frames, the characteristics of a synthesis filter modeling the vocal tract and located at a receiver connected to the transmitter via a low-bit-rate data link, and (2) a multipulse excitation signal intended for said synthesis filter and supplied by a pulse generating circuit which determines the pulse position and amplitudes by successive approximation minimizing the mean squared error between the original speech signal and the synthetic speech signal obtained from the filter, the pulse generating circuit adds to the amplitude of each of the pulses, at the end of the approximation routine, a corrective term that is a function of the value of the partial derivative of the mean squared error taken relative to the amplitude of the pulse under consideration, taken as an independent variable.

Synchronous active noise cancellation with LPC noise synthesizer

A new active noise cancellation scheme has been developed for the purpose of engine noise cancellation. The major difference between the new approach and the conventional two‐microphone adaptive noise cancellation methods is that the new approach does not require the reference signal. Therefore, the problem of acoustical feedback from the canceling speaker to the reference channel is eliminated. In this new scheme, a linear predictive coding (LPC) technique is employed to synthesize an “engine noise,” which is used as a reference signal, and this synthesized noise is processed by an LMS adaptive filter. Simulation results using a set of digitized engine noise data show that the adaptive filter has a faster convergence rate and more than 10‐dB mean‐squared error improvement by using an LPC noise synthesizer in comparison with using an impulse train as input only. In the case that engine speed is varying or the engine rpm is not available, a multipulse excitation synthesizer has been used to generate the synthesized engine noise. The multipulse excitation method has also provided an improvement of 10 dB in mean‐squared error.

Speech message code modifying arrangement

Natural quality and bit-rate for LPC speech synthesis is improved by encoding the LPC residual signal in a prescribed multipulse format formed for each LPC frame. Voiced, unvoiced, and mixed (hiss plus periodic) excitation is inherent. The speaking-rate is changed by adding, deleting, or repeating pitch-periods, and the pitch (intonation) is changed by adding or deleting zeros in the multipulse excitation signal.

Multipulse LPC speech processing arrangement

A multipulse-code approximation of of one frame of a predictive residual signal may lose the frame-to-frame redundancy. Accordingly, pitch redundancy removal during the iterative process of forming the multipulse sequence, rather than frame-to-frame code comparison, provides reduction of excitation signal bit rate, substantially independent of voice pitch. A speech pattern predictive coding arrangement includes forming a prescribed format multipulse excitation signal for each successive time frame of the pattern. The multipulse excitation signal corresponds to the frame predictive residual. The redundancy in the multipulse excitation signal is reduced by forming a signal representative of the similarities between the current frame speech pattern and the speech pattern of preceding frames and removing such similarities from the multipulse excitation signal. Advantageously, the bit rate of the multipulse excitation signal is reduced and the excitation signal is rendered substantially independent of voice pitch.

Evaluation of a glottal ARMA modeling scheme

In many speech analysis/synthesis schemes, the source of excitation for voiced speech is a train of impulses. Although this is a mathematically sound scheme, there is no physiological basis for it. The quality of speech that has been attained due to the introduction of a dynamically varying source, e.g., a parametric source mode, multipulse excitation, etc., has been found to be better than that produced using impulse excitation. In this paper, a pitch synchronous glottal ARMA analysis/synthesis scheme is proposed. A parametric voice source model is discussed. The voice source and vocal tract parameters are simultaneously estimated. The AR and MA orders of the vocal tract model are variable and their values are decided every period depending on whether a resynthesis efficiency threshold is crossed during the analysis/synthesis procedure. This scheme is compared with two other schemes, viz., (1) closed phase LPC analysis/synthesis and (2) robust LPC analysis/synthesis. The superiority of the proposed scheme over the latter two methods is shown in terms of better formant/bandwidth tracking ability and efficiency of resynthesis.

Multi‐pulse excited linear predictive speech coder

A multi-pulse excitation linear-predictive speech coder operates in accordance with an analysis-by-synthesis method for determining the excitation. The coder (10) comprises an LPC-analyzer (11), a multi-phase excitation generator (13), means (12, 14) for forming an error signal representative of the difference between an original speech signal (s(n)) and a synthetic speech signal (s(n)), a filter (15) for perceptually weighting the error signal and means (16) responsive to the weighted error signal (e(n)) for generating pulse parameters controlling the excitation generator (13) so as to minimize a predetermined measure of the weighted error signal. The LPC-parameters and the pulse parameters of the excitation signal (x(n)) are encoded for efficient storage or transmission. The bit capacity required for pulse position encoding of the excitation signal (x(n)) is considerably reduced by arranging the excitation generator (16) for an excitation signal (x(n)) which in each excitation interval (L) consists of a pulse pattern having a grid of a predetermined number (q) of equidstant pulses and by arranging the control means (16) for generating pulse parameters characterizing the grid position (k) relative to the beginning of the excitation interval (L) and the variable amplitudes (bk (j), 1≦j≦q) of the pulses of the grid.

Amplitude optimization and pitch prediction in multipulse coders

Although the multipulse model is conceptually simple, the problem of locating the pulses is computationally complex. The authors discuss the basic multipulse model and describe a procedure to compute the excitation with optimally adjusted amplitudes. The algorithm provides a framework for computing multipulse excitation with varying degrees of optimization and computational complexity. The authors find that speech quality depends on the pulse rate. They also find that for the same quality, female speech requires a higher pulse rate than male speech. The pitch dependence can be reduced and speech quality improved for high-pitched speakers by incorporating long delay prediction in the multipulse model. >

Synthesis of Chinese by rules based on a multipulse excitation model

According to the model of speech production, the characteristic parameter of speech can be divided into two parts: excitation and vocal tract parameters. Atal proposed the multipulse excitation model that can produce high‐quality sythesized speech. This research shows that the intensity, duration, and pitch mode of single syllable Chinese produced by multipulse excitation may be changed when the adaptive method is utilized to process its multipulse sequences and vocal tract parameter. There are about 10 000 Chinese words in common use, but the pronunciation of many words is the same, so that only about 1300 syllables are independent. The Chinese language is a tone language. Each Chinese word is of four pitch modes, and the vocal tract parameter for the four modes of one word is almost the same. Therefore, there are 400 independent vocal tract parameters and 1300 multipulse sequences in Chinese. Based on the above strategy, a new method of sythesizing Chinese by rules has been proposed. The intelligibility and naturalness of the synthesized speech are satisfactory.

Effects of degradation in vowel steady-state and transition on the perception of coded speech

In order to investigate the perceptual importance of the vowel steady state and transition on the perception of coded speech, and to obtain cues for further reducing the bit rate in speech coding methods, coded speech was systematically deteriorated during vowel steady states and transitions. Multipulse excited speech coding (MPC) [T. Araseki and K. Ozawa, Proc. GLOBECOM, 794–798 (1983)] at 16 kb/s was used as a speech coding method. Perceptual experiments were carried out using 18 kinds of meaningless sequences having a Japanese CVCV structure including 5 vowels, 2 fricatives, and 6 stops. The intelligibility of the vowels and consonants was measured. The results showed that the intelligibility for vowels was the same as that of the original 1 kb/s MPC, even when all of the multipulse excitation information during the vowel steady states was deleted and the multipulse information during transition parts was reduced to 13 of the 16 kb/s MPC. The intelligibility began to decrease when the multipulse information during the transitions was reduced to 16 of the 16 kb/s MPC. The results, including the intelligibility of the consonants, will be discussed in detail.

MPE/LTP speech coder for mobile radio application

In this paper, we describe our contribution to the CEPT effort for the definition of a European standard for the cellular mobile radio. We first give an overview of the Multipulse excitation with long term prediction (MPE/LTP) algorithm. Then, we report details on the implementation (constants, quantizing tables) respectively in the analysis part, the error protection/correction, and the synthesis part of the codec. Finally, we report detailed figures about the implementation efficiency (ROM/RAM/Cycles).

Joint estimation of the LPC parameters and the multi-pulse excitation

In this paper, a joint process estimation algorithm is presented that simultaneously estimates the smooth spectral structure of the speech signal and the pulse-like driving function used in Multi-Pulse Linear Predictive Coding (MPLPC). Experimental results indicate that while the resulting excitation function differs from the normal multi-pulse excitation, the difference between the optimized and non-optimized linear predictive coding parameters is minimal in both a subjective and a numerical sense.

Synthesis of natural sounding pitch contours in isolated utterances using hidden Markov models

A novel technique is introduced for characterizing prosodic structure and is used for speech synthesis. The mechanism consists of modeling a set of observations as a probabilistic function of a hidden Markov chain. It uses mixtures of Gaussian continuous probability density functions to represent the essential, perceptually relevant structure of intonation by observing movements of fundamental frequency in monosyllabic words of varying phonetic structure. High-quality speech synthesis, using multipulse excitation, is used to demonstrate the power of the HMM in preserving the naturalness of the intonational meaning, conveyed by the variation of fundamental frequency and duration. The fundamental frequency contours are synthesized using a random number generator from the models, and are imposed on a synthesized prototype word which had the intonation of a low fall. The resulting monosyllabic words with imposed synthesized fundamental frequency contours show a high level of naturalness and are found to be perceptually indistinguishable from the original recordings with the same intonation. The results clearly show the high potential of hidden Markov models as a mechanism for the representation of prosodic structure by naturally capturing its essentials.