• Speech Sciences
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• v.8 no.1
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• pp.143-157
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• 2001

This paper focused on glottal parameters contributing to the perception of loud voices because energy of a voice is not the only effective factor. We used a formant synthesizer to synthesize loud voices. We divided F0 tilt (the tilt of F0 contour), SQ (Speed Quotient), OQ (Open Quotient) and TL (spectral Tilt Level) into three levels to get different combinations with default values for the other synthesizer parameters. Analysis of listening tests indicated that F0 tilt, SQ, OQ and TL in descending order had significant influence on the perception of loud voices. F0 tilt had a far more significant effect than the others. The influence of SQ increased greatly with the exclusion of F0 tilt as a factor. The interaction between parameters was not significant.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.381-382
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• 2006

In this paper, we present an efficient architecture for connected speech recognition that can be efficiently implemented with FPGA. The architecture consists of newly derived two-level dynamic programming (TLDP) that use only bit addition and shift operations. The advantages of this architecture are the spatial efficiency to accommodate more words with limited space and the computational speed from avoiding propagation delays in multiplications. The architecture is highly regular, consisting of identical and simple processing elements with only nearest-neighbor communication, and external communication occurs with the end processing elements.

• Journal of the Korean Society of Manufacturing Process Engineers
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• v.10 no.6
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• pp.33-38
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• 2011

We describe a research about remote control of mobile robot based on voice command in this paper. Through real-time remote control and wireless network capabilities of an unmanned remote-control experiments and Home Security / exercise with an unmanned robot, remote control and voice recognition and voice transmission are possible to transmit on a PC using a microphone to control a robot to pinpoint of the source. Speech recognition can be controlled robot by using a remote control. In this research, speech recognition speed and direction of self-driving robot were controlled by a wireless remote control in order to verify the performance of mobile robot with two drives.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.885-892
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• 1994

This paper presents the Korean text-to-speech (TTS) algorithm with speed and intonation control capability, and describes the development of the Voice message delivery system employing this TTS algorithm. This system allows the Interpersonal Messaging (IPM) Service users of Message Handling System (MHS) to send his/her text messages to user via telephone line using synthetic voice. In the X.400 MHS recommendation, the protocols and service elements are not specified for the voice message delivery system. Thus, we defined access protocol and service elements for Voice Access Unit based on the application program interface for message transfers between X.400 Message Transfer Agent and Voice Access Unit. The system architecture and operations will be provided.

• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.1
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• pp.101-113
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• 1986

In this paper, a real-time pitch detector LPC vocoder as implemented on a high speed digital signal processor, NEC 7720, is described. The pitch detector was based mainly on the SIFT algorithm. The SIFT pitch detector consists primarily of a digital low pass filter, inverse filter, computation of autocorrelation, a peak picker, interpolation, V/UV defcision and a final pitch smoother. In our approach, modification, mainly on the V/UV decision and a final pitch smoother, was made to estimate more accurate pitches. An 16-bit fixed-point aithmatic was employed for all necessary computation and the simulated results were compared with the eye detected pitches obtained from real speech data. The pitch detector occupies 98.8% of the instruction ROM, 37% of the data ROM, and 94% of internal RAM and takes 15.2ms to estimate a pitch when an analysis frame is consisted of 128 sampled speech data. It is observed that the tested results were well agreed with the computer simulation results.

• KIISE Transactions on Computing Practices
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• v.22 no.11
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• pp.591-600
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• 2016

Computing technologies are increasingly applied to most casual human environment networks, as computing technologies are further developed. In addition, the rapidly increasing interest in IoT has led to the wide acceptance of speech recognition as a means of HCI. In this study, we present a novel method for recognizing the Korean vowel 'ㅜ', as a part of a phoneme based Korean speech recognition system. The proposed method involves analyses of bulk indicators calculated in the time domain instead of analysis in the frequency domain, with consequent reduction in the computational cost. Four elementary algorithms for detecting typical waveform patterns of 'ㅜ' using bulk indicators are presented and combined to make final decisions. The experimental results show that the proposed method can achieve 90.1% recognition accuracy, and recognition speed of 0.68 msec per syllable.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2005.11a
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• pp.45-48
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• 2005

In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and reduces the computational complexity. The proposed filter is applied to wavelet based adaptive algorithm. Actually we compared the performance of the proposed algorithm with other algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, We know the proposed algorithm is prefer than the existent algorithm.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2459-2462
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• 2003

In this paper, we Propose a wavelet based adaptive algorithm which improves the convergence speed and reduces computational complexity using the fast running FIR filtering efficiently. We compared the performance of the proposed algorithm with time and frequence domain adaptive algorithm using computer simulation of adaptive noise canceler based on synthesis speech.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.137-140
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• 1995

In the paper, CDMA's vocoder algorithm, QCELP, was analyzed. And, 16-bit programmable DSP core for QCELP was designed. When it is used two MACs in DSP, we can implement low-power DSP and estimate decrease of parameter computation speed. Also, we implemented in FIFO memory using register file to increase the access time of the data. This DSP was designed using logic synthesis tool, COMPASS, by top-down design methodology. Therefore, it is possible to cope with rapid change at mobile communication market.

• Journal of the Korean Institute of Intelligent Systems
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• v.10 no.4
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• pp.324-329
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• 2000

본 논문에서는 잡음환경에서의 음질향상(Speed Ehnacement) 시스템 구현을 목적으로 한다. 이를 위한 적응필터로서 LSM(Least Mean square)알고리즘 FIR필터를 적용한다. 또 정밀 필터로서 다충신경망(MLP, Multi-Layer Perceptorn) 필터를 적용한다. 잡음환경에서의 음성신호 복원 및 음질향상 시스템은 잡음에 의해 왜곡된 음성신호에서 잡음성분만을 제거함으로써 음성신호를 복원하는 시스템이다. 신경망 필터는 오차 역전과 학습 알고리즘에 의해 오차를 최소화 하는 방향으로 필터의 피라미터를 수정한다. 제안한 필터로 잡음환경에서의 음성신호복원 시스템을 구서오하고, 실험을 필터의 성능을 확인한다.