• 대한전자공학회논문지
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• 제15권3호
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• pp.1-6
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• 1978

본 논문에서는 음성신호의 디지탈화와 대역폭축소의 한 방법으로 예측부호화 원리를 사용하는 adaptive differential pulse code modulation(ADPCM)과 adoptive delta modulation(ADM)에 관하여 고찰하였다. ADPCM에서 사용되는 대표적인 적응양자기의 원리를 설명하고 적응예측기의 계수를 얻는 두 방법, 즉 브록해석과 연차해석 방법을 검토하였다. 또한 ADM에서 사용되는 세가지 압신방법(instantaneous, syllabic, hybrid commanding)을 구체적으로 설명하고 그의 성능을 비교하였다. 마지막으로 ADPCM과 ADM을 음성신호의 부호화기로 쓸 때의 성능과 장단점들을 비교 검토하였다.

• 전자공학회논문지A
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• 제30A권1호
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• pp.73-85
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• 1993

This paper presents a design and implementation of CAD(Computer-Aided Design) system with tools and design environments for IC(Intergrated Circuits)design. The CAD system can be easily installed in various sites with limited resources, since most CAD tools and design environments are available in the public-domain and Unix & X Window-based PC-386 and Workstation is used for the hardware platform. In order to improve the flexibility of the CAD system, objects are defined in the context of tools and environments` and object tables are programmed to describe the integration of CAD tools and design environments. During the execution, tool-objects deal with intertool communication and round-robin mechanism to incrementally control the execution of CAD tools. The IC design of LPC(Linear Predictive Coding) Speech Synthesizer is carried out to find out improvements and bugs of the CAD system.

• 정보통신설비학회논문지
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• 제10권2호
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• pp.53-61
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• 2011

This research presents an adaptive fast motion estimation (ME) computation on the stage of uneven multi-hexagon grid search (UMHGS) algorithm included in an unsymmetrical-cross multi-hexagon-grid search (UMHexagonS) in H.264 standard. The proposed adaptive method is based on statistical analysis and previously obtained motion vectors to reduce the computational complexity of ME. For this purpose, the algorithm is decomposed into three processes: skipping, terminating, and reducing search areas. Skipping and terminating are determined by the statistical analysis of the collected minimum SAD (sum of absolute difference) and the search area is constrained by the slope of previously obtained motion vectors. Simulation results show that 13%-23% of ME time can be reduced compared with UMHexagonS, while still maintaining a reasonable PSNR (peak signal-to-noise ratio) and average bitrates.

• The Journal of the Acoustical Society of Korea
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• 제17권4E호
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• pp.17-21
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• 1998

This paper presents an implementation of a multi-channel speech surveillance system over telephone lines using TMS320C31 DSP chips. The incoming speech into each telephone line are first compressed simultaneously in real-time by the popular vector-sum excited linear predictive (VSELP) speech coding algorithm at the rate of 8 Kbps. The compressed steech bit streams are then multiplexed with those of other users. The multiplexed speech bit streams are transferred to the system storage equipments with some other required information so that a system operator can later monitor the stored speech data whenever it is necessary. The host program runs under Microsoft Windows95 for an efficient man-machine interface and a future upgrade-ability. We have confirmed that the overall 64-channel system operates satisfactorily in realtime. We also have checked approximately up to 2,880 total hours of recording capability of the system on a playback module and two removable backup drives.

• 대한전기학회논문지:시스템및제어부문D
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• 제51권2호
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• pp.69-77
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• 2002

Speaker verification can be classified in two categories, text-dependent speaker verification and text-independent speaker verification. In this paper, we discuss text-dependent speaker verification. Text-dependent speaker verification system determines whether the sound characteristics of the speaker are equal to those of the specific person or not. In this paper we obtain the speaker data using a sound card in various noisy conditions, apply a new Hidden LMS (Least Mean Square) adaptive algorithm to it, and extract LPC (Linear Predictive Coding)-cepstrum coefficients as feature vectors. Finally, we use a competitive learning neural network for speaker verification. The proposed hidden LMS adaptive filter using a neural network reduces noise and enhances features in various noisy conditions. We construct a separate neural network for each speaker, which makes it unnecessary to train the whole network for a new added speaker and makes the system expansion easy. We experimentally prove that the proposed method improves the speaker verification performance.

• 음성과학
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• 제11권2호
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• pp.227-235
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• 2004

In this paper, we proposed a new practical implementation method of the lattice transversal joint (LTJ) adaptive filter using speech codec's information. And it was applied to the adaptive echo cancellation problem to verify the efficiency of the proposed method. Realtime implementation of the LTJ adaptive filter is very difficult due to high computational complexity for the filter coefficients compensation. However, in case of using speech codec, complexity can be reduced since linear predictive coding (LPC) coefficients are updated each frame or sub-frame instead of every sample. Furthermore, LPC coefficients can be acquired from speech decoder and transformed to the reflection coefficients. Therefore, the computational complexity for updates of the reflection coefficients can be reduced. The effectiveness of the proposed LTJ adaptive filter was verified by the experiments about convergence and tracking performance of the adaptive echo canceller.

• 대한후두음성언어의학회지
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• 제19권2호
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• pp.101-106
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• 2008

The speech signal has been put into a form suitable for storage and analysis by computer, several different operation can be performed. Filtering, sampling and quantization are the basic operation in digiting a speech signal. The waveform can be displayed, measured and even edited, and spectra can be computed using methods such as the Fast Fourier Transform (FFT), Linear predictive Coding (LPC), Cepstrum and filtering. The digitized signal also can be used to generate spectrograms. The spectrograph provide major advantages to the study of speech. So, author introduces the basic techniques for the acoustic recording, digital signal processing and the principles of spectrum and spectrogram.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1988년도 전기.전자공학 학술대회 논문집
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• pp.33-36
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• 1988

In this paper, the starting point of affricate is detected from the first predictor coefficient of a 12-pole linear predictive coding (LPC) analysis and phonemic segmentation is done through measuring short time energy and zero crossing rate. By this segmentation method, the duration of an aspirate can be mearsured in order to detect an aspirate or not.

• 정보저장시스템학회논문집
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• 제6권1호
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• pp.18-24
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• 2010

The performance of the speech recognition mainly depends on uncertain factors such as speaker's conditions and environmental effects. The present study deals with the speech recognition of a number of multi-syllable isolated Korean words using soft computing techniques such as back-propagation neural network, fuzzy inference system, and fuzzy neural network. Feature patterns for the speech recognition are analyzed with 12th order thirty frames that are normalized by the linear predictive coding and Cepstrums. Using four models of speech recognizer, actual experiments for both single-speakers and multiple-speakers are conducted. Through this study, the recognizers of combined fuzzy logic and back-propagation neural network and fuzzy neural network show the better performance in identifying the speech recognition.

• 한국음향학회지
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• 제1권1호
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• pp.45-53
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• 1982

A robust linear predictive coding method that can be used in noisy as well as quiet environment has been studied. In this method, noise autocorrelation coeffieients are first obtained and updated during nonspeech periods. Then, the effect of additive noise in the input speech is removed by subtracting values of the noise autocorrelation coefficients of corrupted speech in the course of computation of linear prediction coefficients. When signal-to-noise ratio of the input speech ranges from 0 to 10 dB, a performance improvement of about 5 dB can be gained by using this method. The proposed method is computationally very efficient and requires a small storage area.