• The Journal of the Acoustical Society of Korea
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• v.14 no.2
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• pp.50-61
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• 1995

A time scale modification method yielding rate-modified speech while conserving the characteristic of speech was implemented in real-time using a goneral purpose digital signal processor. Time scale modification changed pronunciation speed only, producing a time difference between the input signal and the modified signal, making it impossible to implement it in real-time. In this thesis, a system was implemented to remove the time difference between the input and modified signals. Speech signals slowed down or speeded up by a physical time scale modification method, such as adjusting the motor speed of the cassett tape recorder, was used as the input signal. Physical modification that controled only the inter speed of the cassette tape player distorted the pitch period of the original speech. In this study, a real-time system was implemented so that the pitch-distorted speech was reconstructed back to the original by fractional sampling pitch shifting using an FIR filter, and this signal was time scale modified to match the cassette tape recorder motor speed using SOLA time-scale medification. In experiments using speech signals medifiedby the proposed method, results obtained using a 16-bit resolution ADSP2101 processor and using computer simulations employing floating point operations showed about the same average frame signal-to-noise ratio of about 20 dB.

• Journal of Korea Multimedia Society
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• v.16 no.9
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• pp.1005-1017
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• 2013

Voice activity detection is very important process that voice activity separated form noisy speech signal for speech enhance. Over the past few years, many studies have been made on voice activity detection, but it has poor performance in low signal to noise ratio environment or fickle noise such as car noise. In this paper, it proposed new voice activity detection algorithm using ensemble variance based on wavelet band entropy and soft thresholding method. We conduct a survey in a lot of signal to noise ratio environment of car noise to evaluate performance of the proposed algorithm and confirmed performance of the proposed algorithm.

• The Proceeding of the Korean Institute of Electromagnetic Engineering and Science
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• v.5 no.1
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• pp.100-110
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• 1994

레이다 신호는 대표적인 전자파 신호로서 주변환경에 따라 시간, 주파수, 공간 영역에서 고유한 신호특성을 가지고 있으며, 신호처리 기법도 다양하다. 본 논문에서는 먼저 레이다를 위한 전파 신호처리 의정의와 필요성을 언급한뒤, 레이다 신호환경 특성을 살펴보고 신호처리를 위한 신호의 시간 및 스펙트럼 특성에 대해 기술하였다. 그리고, 신호특성에 적합한 신호처리기의 구현을 위해 레이다 신호처리에 관 련된 주요 기법에 대해 개괄적으로 설명하였다. 레이다 신호처리 분야는 일반적으로 잘 알려진 음성이 나 영상신호처리 분야와 달리 고유한 알고리듬과 구조가 요구된다. 신호처리기법으로서 레이다 파형설 계, 해상도 모호성, 펄스압축, 클러터제거, 도플러처리, 일정오경보탐지, 클러터 지도, 표적군 형성/ 추출, 표적식별, 레이다영상기법, 적응배열처리 등에 관해 개괄적으로 설명하였다. 레이다 선호처리 기술은 "스마트"한 레이다를 위한 두뇌 역할을 하기때문에 그 필요성과 중요성이 증가하고 있다. 그러나, 고속, 대용량의 신호를 주어진 빔 주사시간동안에 실시간으로 처리하여 표적 정보를 추출해야 하기 때문에 아직도 상용 프로세서의 속도 한계내에서 알고리듬의 수행에 다소 제약을 받고 있으나, 최근 디지탈 신호처리 전용의 고속 칩의 출현으로 많은 발전을 가져오고 있다. 끝으로, 향후 레이다 신호처리 발전 추세와 응용분야에 대해 살펴보았다. 응용분야는 군수 및 민수용의 겸용 파급효과가 매우 크고, 군용의 대공탐색 및 조기경보, 전장감시뿐만 아니라 전투기 탑재용으로 필수적이며, 특히 민수용의 공 항, 항공기, 선박, 위성 등 매우 다양하다. 최근 발전추세에 따른 기술로서 다중모드 신호처리, 고집적 회로기술, 적응배열, 디지탈 빔형성, 적응성, 고분해능 및 방향성, 표적식별, 다차원 신호처리에 대해 언급 하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.8
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• pp.1665-1674
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• 2011

In this paper, we proposed an algorithm for speech enhancement of speeches corrupted by colored noise. When there is no correlation between colored noise and speech signal, the colored noise turns into white noise through whitening transformation. This transformed signal has been applied to the generalized subspace approach for speech enhancement. The speech spectral distortion, produced by the whitening transformation as pre-processing, has been restored by using the inverse whitening transformation as post-processing of the proposed algorithm. The performance of the proposed algorithm for speech enhancement has been confirmed by computer simulation. The colored noises used in this experiment were car noise and multi-talker babble. It is confirmed that the proposed algorithm shows better performance from SNR and SSD viewpoint over the previous approach with the data from the AURORA and TIMIT data base.

• Journal of Digital Convergence
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• v.16 no.12
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• pp.459-466
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• 2018

Gendered innovations is a term used by policy makers and academics to refer the process of creating better research and development (R&D) for both men and women. In this paper, we analyze the literatures in image and speech signal processing that can be used in ICT, examine the importance of gendered innovations through case study. Therefore the latest domestic and foreign literature related to image and speech signal processing based on gender research is searched and a total of 9 papers are selected. In terms of gender analysis, research subjects, research environment, and research design are examined separately. Especially, through the case analysis of algorithms of the elderly voice signal processing, machine learning, machine translation technology, and facial gender recognition technology, we found that there is gender bias in existing algorithms, and which leads to gender analysis is required. We also propose a gendered innovations method integrating sex and gender analysis in algorithm development. Gendered innovations in ICT can contribute to the creation of new markets by developing products and services that reflect the needs of both men and women.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.1474-1475
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• 2013

부항은 열 또는 음압(陰壓)장치에 의하여 부항단지 안에 음압을 조성하여 피부에 붙임으로써 피를 뽑거나 울혈(鬱血)을 일으키며 물리적 자극을 주어 병을 치료한다. 부항으로 얻어지는 물리적인 자극은 혈액순환을 촉진하고, 죽은피를 빼냄으로써 혈관을 자극하고 그로인해 다양한 효과를 얻는다. 따라서 본 논문에서는 신장에 해당하는 명문혈을 자극하여 신장과 관련된 음성분석 요소의 변화를 측정하였다. 이를 위해 신장에 이상이 없는 피실험자 10명을 선정하고 신장에 해당하는 명문혈을 자극하기 전과 후의 음성을 수집하였다. 실험은 음성분석 요소 중 신장과 관련된 1 Formant Bandwidth를 적용하여 신장 명문혈 자극 전과 후의 변화를 측정, 분석하였다. 실험 결과, 90%의 피실험자가 값이 감소하는 현상을 보였으며, 이를 통해 명문혈 자극에 따른 신장과 음성신호와의 상관성을 분석할 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.395-401
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• 2005

In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.12
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• pp.1321-1328
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• 2015

A radio is a half-duplex voice communication method using the PTT(: Push To Talk), occupy a single line calls during transmission. As an interface between the telephone and the radio, UHF and VHF, for voice communication between the different heterogeneous devices, A device automatically switches between the two devices is required. Therefore, in accordance with the performance of the voice switching apparatus for detecting a voice to be transmitted from an input signal, loss of the audio signal to be transmitted is subjected to Significant influence. Conventional method has the problem responding to noise by setting the level through simple means of amplitude of input signal, in other words, the energy level of the input signal. This paper, by using the audio signal processing techniques, this discriminated what the voice is among the input signal and substantiated a device for the automatic voice transmission between heterogeneous devices. With this proposal, I was confirmed of improvement of performance in the automatic voice switching device, could perform loss-less transmission of voice between heterogeneous devices.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.36-40
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• 1997

In speech signal processing, it is necessary and important to detect exactly the pitch. The algorithms of pitch extraction which have been proposed until now are difficult exactly pitches over wide range speech signals. In this paper, thus, we proposed a new pitch detection algorithm that finds the fundamental period of speech signal in the residual signal quantized by the uniform quantizer as PCM. The proposed method shows little gross error of average 0.25% for clean speech and average 3.39% for SNR of 0dB. It also achieves results of the pitch contours, improving the accuracy of pitch detection in transient phonemes and noise environments.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.3
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• pp.61-67
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• 2001

In this paper, a method that apply LP algorithm to image for speech recognition is suggested, using both speech and image information for recogniton of korean numeral speech. The input speech signal is pre-emphasized with parameter value 0.95, analyzed for B th LP coefficients using Hamming window, autocorrelation and Levinson-Durbin algorithm. Also, a gray image signal is analyzed for 2-dimensional LP coefficients using autocorrelation and Levinson-Durbin algorithm like speech. These parameters are used for input parameters of neural network using back-propagation algorithm. The recognition experiment was carried out at each noise level, three numeral speechs, '3','5', and '9' were enhanced. Thus, in case of recognizing speech with 2-dimensional LP parameters, it results in a high recognition rate, a low parameter size, and a simple algorithm with no additional feature extraction algorithm.