An ASCII encoding of Korean has been developed for extended phonetic transcription of the Speech Assessment Methods Phonetic Alphabet (SAMPA). SAMPA is a machine-readable phonetic alphabet used for multilingual computing. It has been developed since 1987 and extended to more than twenty languages. The motivating factor for creating Korean SAMPA (K-SAMPA) is to label Korean speech for a multilingual corpus or to transcribe native language (Ll) interfered pronunciation of a second language learner for bilingual education. Korean SAMPA represents each Korean allophone with a particular SAMPA symbol. Sounds that closely resemble it are represented by the same symbol, regardless of the language they are uttered in. Each of its symbols represents a speech sound that is spectrally and temporally so distinct as to be perceptually different when the components are heard in isolation. Each type of sound has a separate IPA-like designation. Korean SAMPA is superior to other transcription systems with similar objectives. It describes better the cross-linguistic sound quality of Korean than the official Romanization system, proclaimed by the Korean government in July 2000, because it uses an internationally shared phonetic alphabet. It is also phonetically more accurate than the official Romanization in that it dispenses with orthographic adjustments. It is also more convenient for computing than the International Phonetic Alphabet (IPA) because it consists of the symbols on a standard keyboard. This paper demonstrates how the Korean SAMPA can express allophonic details and prosodic features by adopting the transcription conventions of the extended SAMPA (X-SAMPA) and the prosodic SAMPA(SAMPROSA).

To develop the code of predicting flow-field and aeroacoustic noise by an electrical cable, a combined CFD-Acoustic analogy approach is selected. The two dimensional, unsteady and incompressible Reynolds-averaged Navier-Stokes solver with κ-ω and κ-ω SST turbulence modeling is used to calculate the near flow-field around an electric cable. Near-field results are then coupled with two-dimensional Curle's integral formulation based upon Lighthill's acoustic analogy with the assumption of acoustic compactness. To validate this code, numerical results are compared with experimental data for a circular cylinder. The simulation shows an overprediction on acoustic amplitudes, but overally speaking, the spectrum pattern of sound pressure agrees well with experiment within an acceptable amount of error. In addition, a few cross-sections of the cable were selected and tested with each other in terms of drag and radiated noise

This paper describes a fast harmonic estimation, referred to as Delta Adjustment (DA), using a low resolution pitch. The presented DA method is based on modification of the Generalized Dual Excitation (GDE) technique[1] which was proposed to improve speech enhancement performance. We introduce the GDE technique and modify it to be suitable for low bit rate harmonic coding that uses only an integer pitch estimate. Unlike the GDE, the DA matches a frequency-warped version of the original spectrum that conforms to a fixed pitch at all harmonic bands. In addition, complexity and performance of the presented method are described in comparison with those of the conventional Fractional Pitch (FP) based harmonic estimation. Experimental results showed that the DA algorithm significantly reduces the complexity of the FP method while maintaining the performance.

A large-scale synthesis database for a unit selection based synthesis method usually retains redundant synthesis unit instances, which are useless to the synthetic speech quality. In this paper, to eliminate those instances from the synthesis database, we proposed a new pruning method called weighted vector quantization (WVQ). The WVQ reflects relative importance of each synthesis unit instance when clustering the similar instances using vector quantization (VQ) technique. The proposed method was compared with two conventional pruning methods through the objective and subjective evaluations of the synthetic speech quality: one to simply limit maximum number of instance, and the other based on normal VQ-based clustering. The proposed method showed the best performance under 50% reduction rates. Over 50% of reduction rates, the synthetic speech quality is not seriously but perceptibly degraded. Using the proposed method, the synthesis database can be efficiently reduced without serious degradation of the synthetic speech quality.

We investigate the numerical instability of linear prediction for discrete harmonic spectra of audio signals. It is identified that the eigenvalue spread is very large when discrete harmonic spectra are confined only in a lower part of the entire signal bandwidth. A simple method that redefines the sampling frequency and associate harmonic frequencies is proposed to improve the numerical stability. Simulation results using real audio signals indicate its superior stabilizing ability and improved accuracy in the discrete spectral estimation for both LP and DAP.

This paper examines the dispersion relation governing the wave propagation on cylindrical shells. The assumption of thin shells allows the dispersion relation to be separated into three relations related to the propagation of flexural waves and two types of membrane waves. Those relations are used to identify the characteristics of the wave number curves. The dispersion relation provides two and three closed wave number curves below and above the ring frequency. Above the ring frequency three wave number curves are clearly identified to be those of flexural, shear and longitudinal waves, respectively. Below the ring frequency, the characteristics of two wave number curves are identified with dependence of the direction of wave propagation.

This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.