• The Journal of the Korea institute of electronic communication sciences
• /
• v.6 no.2
• /
• pp.186-192
• /
• 2011

In this paper, we propose a new procedure for improvement of the linear predictive coding. To reduce the error power incurred by the coding, we interchanged the order of the two procedures of windowing on the signal and linear prediction. This scheme corresponds to LPC extraction with windowed autocorrelation. The proposed method requires more calculational time because it necessitates matrix inversion on more parameters than the conventional technique where an efficient Levinson-Durbin recursive procedure is applicable with smaller parameters. Experimental test over various speech phonemes showed, however, that our procedure yields about 5 % less power distortion compared to the conventional technique. Consequently, the proposed method in this paper is thought to be preferable to the conventional technique as far as the fidelity is concerned. In a separate study of speaker-dependent speech recognition test for 50 isolated words pronounced by 40 people, our approach yielded better performance too.

• Proceedings of the KSPS conference
• /
• 1996.10a
• /
• pp.207-207
• /
• 1996

The neutral tone is one of the most important distinguishing features in Beijing Mandarin, but there are two completely different views on its linguistic function: a special tone(Xu, 1980) versus weak stress(Chao, 1968). In this paper, the acoustic manifestation of the neutral tone will be explored to show that it is closely related to weak stress. 122 disyllabic words in which the second syllable carries the neutral tone, including 22 stress pairs, were uttered by a native male speaker of Beijing dialect and analysed by Kay Digital Sonagraph 5500-1. The results of the acoustic analysis are presented as follows: 1) The first two formants of the medial and the syllabic vowel moves towards that of central vowel with a greater magnitude in the syllable with the neutral tone than in the syllable with any of the four normal tones. Also the vowel ending, and nasal coda /n/ and / / in the syllable with the neutral tone tends to be deleted. 2) In the syllables with the neutral tone, there are strong carryover coarticulations between the medial and syllabic vowel and the preceding unvoiced consonant. In general, the vowel is affected to move towards the position of the central vowel with more greater magnitude by coronal consonant than by labial or velar consonant. 3) In the syllable with the neutral tone, when and only when it precedes a syllable with tone-4, the high vowel following [f], [ts'], [s], [ts'], [s], [tc'] or [c] tends to be voiceless. 4) It can be seen from the acoustical results of 22 stress pairs that the duration of the syllable with the neutral tone is on the average reduced to 55% of that of the syllable with the four normal tones, and the duration of the final in the syllable with neutral tone is on the average reduced to 45% of that of the final in the syllable with the four normal tones(Lin & Yan 1980). 5) The FO contour of the neutral tone is highly dependent on the preceding normal tone(Lin & Yan 1993). For a number of languages it has been found that the vowel space is reduced as the level of stress placed upon the vowel is reduced(Nord 1986). Therefore we reach the conclusion that the syllable with neutral tone is related to weak stress(Lin & Yan 1990). The neutral tone is not a special tone because the preceding normal tone.

• Journal of Institute of Control, Robotics and Systems
• /
• v.13 no.8
• /
• pp.726-734
• /
• 2007

Generally, a mobile robot is moved by original input programs. However, it is very hard for a non-expert to change the program generating the moving path of a mobile robot, because he doesn't know almost the teaching command and operating method for driving the robot. Therefore, the teaching method with speech command for a handicapped person without hands or a non-expert without an expert knowledge to generate the path is required gradually. In this study, for easily teaching the moving path of the autonomous mobile robot, the autonomous mobile robot with the function of speech recognition is developed. The use of human voice as the teaching method provides more convenient user-interface for mobile robot. To implement the teaching function, the designed robot system is composed of three separated control modules, which are speech preprocessing module, DC servo motor control module, and main control module. In this study, we design and implement a speaker dependent isolated word recognition system for creating moving path of an autonomous mobile robot in the unknown environment. The system uses word-level Hidden Markov Models(HMM) for designated command vocabularies to control a mobile robot, and it has postprocessing by neural network according to the condition based on confidence score. As the spectral analysis method, we use a filter-bank analysis model to extract of features of the voice. The proposed word recognition system is tested using 33 Korean words for control of the mobile robot navigation, and we also evaluate the performance of navigation of a mobile robot using only voice command.

• The Journal of the Acoustical Society of Korea
• /
• v.24 no.4
• /
• pp.186-193
• /
• 2005

The Terrestrial Digital Multimedia Broadcasting (T-DMB) system supplies high quality audio comparable with VCD in 7 inch display and high quality audio comparable CD at the mobile reception environment T-DMB will launch commercial service at the middle of 2005. However the bandwidth for audio data and the number of channels are restricted to 128 kbps and 2 respectively in the current T-DMB standard because of the limitation of available bandwidth for multimedia data. This Paper Proposes a novel media processor structure for providing multi-channel audio contents oyer T-DMB system allowing backward compatibility with the legacy T-DMB receiver. Furthermore. we also Propose an adaptive receiver structure to supply optimal audio contents on various speaker configuration in T-DMB receiver. To provide multi-channel audio contents allowing backward comaptilbity with the legacy T-DMB receiver, the additional data for multi-channel audio are defined as a dependent stream of main audio stream. The OD strucure for control an additional multi-channel audio elementary stream is proposed without changing the BIFS of the legacy T-DMB system.

• Journal of Digital Convergence
• /
• v.16 no.11
• /
• pp.335-340
• /
• 2018

Commercialized speech recognition systems that have an accuracy recognition rates are used a learning model from a type of speaker dependent isolated data. However, it has a problem that shows a decrease in the speech recognition performance according to the quantity of data in noise environments. In this paper, we proposed the vector quantization based speech recognition performance improvement using maximum log likelihood in Gaussian distribution. The proposed method is the best learning model configuration method for increasing the accuracy of speech recognition for similar speech using the vector quantization and Maximum Log Likelihood with speech characteristic extraction method. It is used a method of extracting a speech feature based on the hidden markov model. It can improve the accuracy of inaccurate speech model for speech models been produced at the existing system with the use of the proposed system may constitute a robust model for speech recognition. The proposed method shows the improved recognition accuracy in a speech recognition system.

• The Journal of the Acoustical Society of Korea
• /
• v.28 no.3
• /
• pp.290-297
• /
• 2009

This paper presents an improved adaptive multi-rate wideband (AMR-WB) algorithm for the efficient Text-To-Speech (TTS) database compression. The proposed algorithm includes unnecessary common bit-stream (CBS) removal and parameter delta coding combined with speaker-dependent huffman coding to reduce the required bit-rate without any quality degradation. We also propose lossy coding schemes to produce the maximum bit-rate reduction with negligible quality degradation. The proposed lossless algorithm including CBS removal can reduce bit-rate by 12.40% without quality degradation compared with the 12.65 kbps AMR-WB mode. The proposed lossy algorithm can reduce bit-rate by 20.00% with 0.12 PESQ degradation.

• Journal of the Korean Society for Nondestructive Testing
• /
• v.29 no.5
• /
• pp.436-441
• /
• 2009

A fiber optic sensor is prospective to be applied to structural health monitoring. Especially, a fiber Bragg grating(FBG) sensor is one of the most popular sensors for the structural health monitoring. The FBG sensor has several demodulation systems for tracking the shift of the Bragg wavelength. The dynamic bandwidth is dependent on the demodulation system. In this paper, the sensing mechanism is that the slope of the optical spectrum of FBG could be used as its sensitivity when the tunable laser shot the monochromatic laser wavelength at the highest slope point. In this technique, the high sensitivity is guaranteed even though the sensing range is limited. In an example of the application, the composite plate embedding a FBG sensor was manufactured by using an autoclave method and the above sensing mechanism was applied to the composite plate. Firstly, the natural frequencies of the plate were successfully measured by the FBG sensor during the impact hammer test. Secondly, a high-power speaker was used to force the plate to be vibrated at the specific frequency that was one of the natural frequencies. During the shaking, the FBG sensor measures the dynamic characteristics and ESPI was also used to measure the mode shape. From the two dynamic tests, the availability of the FBG sensor system and the ESPI was proven as a technique for measuring the dynamic characteristics of composite structure.

• The Journal of the Acoustical Society of Korea
• /
• v.14 no.2
• /
• pp.101-112
• /
• 1995

In order to construct a feature map-based phoneme classification system for speech recognition, two procedures are usually required. One is clustering and the other is labeling. In this paper, we present a phoneme classification system based on the Kohonen's Self-Organizing Feature Map (SOFM) for clusterer and labeler. It is known that the SOFM performs self-organizing process by which optimal local topographical mapping of the signal space and yields a reasonably high accuracy in recognition tasks. Consequently, SOFM can effectively be applied to the recognition of phonemes. Besides to improve the performance of the phoneme classification system, we propose the learning algorithm combined with the classical K-mans clustering algorithm in fine-tuning stage. In order to evaluate the performance of the proposed phoneme classification algorithm, we first use totaly 43 phonemes which construct six intra-class feature maps for six different phoneme classes. From the speaker-dependent phoneme classification tests using these six feature maps, we obtain recognition rate of $87.2\%$ and confirm that the proposed algorithm is an efficient method for improvement of recognition performance and convergence speed.

• Journal of the Korean Institute of Intelligent Systems
• /
• v.16 no.1
• /
• pp.49-54
• /
• 2006

The recent fingerprint recognition system has unstable factors, such as copy of fingerprint patterns and hacking of fingerprint feature point, which mali cause significant system error. Thus, in this research, we used the fingerprint as the main recognition device and then implemented the multi-biometric recognition system in serial using the speech recognition which has been widely used recently. As a multi-biometric recognition system, once the speech is successfully recognized, the fingerprint recognition process is run. In addition, speaker-dependent DTW(Dynamic Time Warping) algorithm is used among existing speech recognition algorithms (VQ, DTW, HMM, NN) for effective real-time process while KSOM (Kohonen Self-Organizing feature Map) algorithm, which is the artificial intelligence method, is applied for the fingerprint recognition system because of its calculation amount. The experiment of multi-biometric recognition system implemented in this research showed 2 to $7\%$ lower FRR (False Rejection Ratio) than single recognition systems using each fingerprints or voice, but zero FAR (False Acceptance Ratio), which is the most important factor in the recognition system. Moreover, there is almost no difference in the recognition time(average 1.5 seconds) comparing with other existing single biometric recognition systems; therefore, it is proved that the multi-biometric recognition system implemented is more efficient security system than single recognition systems based on various experiments.