The Jing is a traditional Korean percussion instrument which plays a major role in Korean folk music. The distingishing feature of this instrument is its unique, long lasting low tone timbre. In this paper, we investigated the vibro-acoustic characteristics of the Jing. Our attention was focused mainly on findings out the physical variables that determine its unique sound. By understanding the way in which the Jing is manufactured, we were able to realize that the unique manufacturing and especially the tuning process by expert craftsman is responsible for the peculiar timbre the Jing produces. The experimental methods implemented to analyzer the Jing were planar acoustic holography and direct measurements by accelerometers. The results from the holographic method and the direct measurements were in good agreement. It turned out that unlike most percussion instruments which have inharmonic partials, the Jing has harmonic partials which are responsible for its unique low-tone timbre. From the holographic representations of the modes, it is clear that the antinodes are located in the center of the Jing which is coincident with the typical striking location. In addition, intensity maps were constructed so that the specific acoustic energy flow can be visualized. It was also interesting to see the the circulation of energy intensity which corresponds to the rotating mode of the Jing.

The future teleconferencing systems will need an appropriate system which controls properly the acoustic echo for the convenient communication. The conventional acoustic echo cancellation algorithms involve large adaptive filters identifying the impulse response of the echo path. The use of adaptive IIR filters appears to be a reasonable way to reduce computational complexity. Effective cancellation of acoustic echo presented in teleconferencing system requires that adaptive filters have a rapid convergence speed. One of the main problems of acoustic echo cancellation techniques is that the convergence properties degrade for an highly correlated signal input such as speech signals. By the way, the introduction of linear prediction filers onto the structure of the acoustic echo cancellation represents one approach to decorrelate the speech signal. And variable step-size LMS algorithm improves the convergence speed through a little increasing of computational complexity. In this paper, we applied these two methods to the acoustic echo canceller(AEC) and showed that these methods have better performances than the conventional AEC.

We propose to use transverse waves instead of longitudinal waves in a scanning tomographic acoustic microscope (STAM) and new type of multiple-transducer with the functions of muliple-angle and multiple-frequency tomography. Proposed multiple-transducer scheme has three insonification angles and three resonance frequencies in order to operate in the transverse wave mode and multiple-angle and multiple frequency tomography for the STAM. In order to evaluate the performance of proposed transducer scheme we have simulated tomographic reconstruction with back-and-forth propagation(BFP) algorithm. Simulation results showed proposed multiple-transducer scheme is capable of obtaining good resolution with transverse wave mode and multiple-frequency tomography. It is also showed that proposed scheme is an efficient rotation scheme by proportion to the number of projections.

This paper proposes a new rejection algorithm which distinguishes unregistered spoken words(or non-keywords) from registered vocabulary. Two kinds of garbage models are employed in this design ; the original garbage model and a new word garbage model. The original garbage model collects all non-keyword patterns where the new word garbage model collects patterns classified by recognizing each non-keyword pattern with registered vocabulary. These two types of garbage models work together to make a robust reject decision. The first stage of processing is the classification of an input pattern through the original garbage model. In the event that the first stage of processing is ambiguous, the new word dependent garbage model is used to classify thye input pattern as either a registered or non-registered word. This paper shows the efficiency of the new word dependent garbage model. A Dynamic Multisection method is used to test the performance of the algorithm. Results of this experiment show that the proposed algorithm performs at a higher level than that of the original garbage model.

This paper proposes a isolated speech recognition method of Korean digits using a TDNN(Time Delay Neural Network) which is able to recognizc time-varying speech properties. We also make an investigation of effect on network parameter of TDNN ; hidden layers and time-delays. TDNNs in our experiments consist of 2 and 3 hidden layers and have several time-delays. From experiment result, TDNN structure which has 2 hidden-layers, gives a good result for speech recognition of Korean digits. Mis-recognition by time-delays can be improved by changing TDNN structures and mis-recognition separated from time-delays can be improved by changing input patterns.

The objective of the paper is to compare the computational complexity of five algorithms for computing time-domain second-order Volterra filter outputs in terms of number of real multiplication and addition operations required for implementation. This study shows that if the filter memory length is greater that or equal to 16, the fast algorithm using the overlap-save method and the frequency-domain symmetry properties of the quadratic coefficients is the most efficient among the algorithms investigated in this paper, When the filter memory length is less than 16, the algorithm using the time-domain symmetry properties is better than any other algorithm.

This paper presents two results on intersymbol interferences in baseband digital communication over an additive white Gaussian noise channel-the interferences due to mismatched square-root raised-cosine filters, in which the filters have different roll-off factors, and / or due to sampling-time jitter. The result for the mismatched filters is that even the jitter-free sampling causes intersymbol interference and it is negligibly small for a wide range of signal-to-noise ratio up to 10dB, for the roll-off factor ranging from 0.2 to 0.5, the mismatch loss being within 0.1dB from the optimum at around 10-6 .For jitter interference an approximation formula for the bit error probability is derived in case of the matched filters, which shows how the roll-off factors and the amount of jitter affect the system performance. The formula is reasonably accurate.

This paper proposes a MSHMM(Multi-Section Hidden Markov Model) recognition method based on Fuzzy Concept, as a method on the speech recognition of speaker-independent. In this recognition method, training data are divided into several section and multi-observation sequences given proper probabilities by fuzzy rule according to order of short distance from MSVQ codebook per each section are obtained. Thereafter, the HMM per each section using this multi-observation sequences is generated, and in case of recognition, a word that has the most highest probability is selected as a recognized word. In this paper, other experiments to compare with the results of these experiments are implemented by the various conventional recognition methods(DP, MSVQ, DMS, general HMM) under the same data. Through results of all-round experiment, it is proved that the proposed MSHMM based on fuzzy concept is superior to DP method, MSVQ method, DMS model and general HMM model in recognition rate and computational time, and does not decreases recognition rate as 92.91% in spite of increment of speaker number.

The effect of pitch on tone durations was studied in several poeces of music. In previous work it was found that higher pitched sounds, of a given duration, were perceived to be of longer duration. That work however, was not undertaken in a musical context. The present work was done to determine whether the previously observed trend was also apparent in musical performances. Several measurements were undertaken. One of these involved analysing the recorded performance duration of notes in several performances of the Mozart Flute Quartet in A-mojor, K.298. Performed durations were also obtained from a Bach Flute Sonata and a Brahms Clarinet Quintet. An analysis of the musical performances revealed that while there were individual differences and exceptions in notes, the musicians tended to play higher notes significantly shortened.