• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2009년도 정보 및 제어 심포지움 논문집
• /
• pp.37-39
• /
• 2009

In the research of speech recognition, locating the beginning and end of a speech utterance in a background of noise is of great importance. Since the background noise presenting to record will introduce disturbance while we just want to get the stationary parameters to represent the corresponding speech section, in particular, a major source of error in automatic recognition system of isolated words is the inaccurate detection of beginning and ending boundaries of test and reference templates, thus we must find potent method to remove the unnecessary regions of a speech signal. The conventional methods for speech endpoint detection are based on two simple time-domain measurements - short-time energy, and short-time zero-crossing rate, which couldn't guarantee the precise results if in the low signal-to-noise ratio environments. This paper proposes a novel approach that finds the Lyapunov exponent of time-domain waveform. This proposed method has no use for obtaining the frequency-domain parameters for endpoint detection process, e.g. Mel-Scale Features, which have been introduced in other paper. Comparing with the conventional methods based on short-time energy and short-time zero-crossing rate, the novel approach based on time-domain Lyapunov Exponents(LEs) is low complexity and suitable for Digital Isolated Word Recognition System.

• 대한전자공학회논문지
• /
• 제24권2호
• /
• pp.196-200
• /
• 1987

The following methods are proposed for implementing digital dual tone multi-frequency (DTMF) receiver: using infinite impulse response(IIR) digital filters, period-counting algorithm, discrete Fourier transform(DFT), and fast Fourier transform(FFT)[2]. The PARCOR(Partical Correlation) analysis method which has been widly used in the speech signal processing area is applied to the dual tone multi-frequency(DTMF) signal detection. This method is easy to implement digitally and stronger to digit simulation of speech than any other methods proposed up to date. Since sampling rate of 4KHz is used in the DTMF receiver for the detection of input DTMF signal originally sampled at 8KHz, it effects two times higher multiplexing efficiency.

• 대한전기학회:학술대회논문집
• /
• 대한전기학회 1988년도 전기.전자공학 학술대회 논문집
• /
• pp.198-202
• /
• 1988

In digital mobile communications received speech data are affected by burst errors as well as random errors. To overcome these errors we propose a bit-selective forward error correction scheme for the speech data which is sub-band coded at 13 kbps and transmitted over a 16 kbps channel. For a few error correcting codes the signal-to-noise ratio of error-corrected speech is obtained and compared through the simulation of mobile communication channels.

• 대한음성학회지:말소리
• /
• 제65호
• /
• pp.143-152
• /
• 2008

This paper focuses on the real time implementation of an adaptive multi-rate (AMR) speech codec, that is a standard speech codec of IMT-2000, using the TMS320VC5510. The series of TMS320VC55x is a 16-bit fixed-point digital signal processor (DSP) having low power consumption for the use of mobile communications by Texas Instruments (TI) corporation. After we analyze the AMR algorithm and source code as well as the structure and I/O of 7MS320VC55x, we carry out optimizing the programs for real time implementation. The implemented AMR speech codec uses 55.2 kbyte for the program memory and 98.3 kbyte for the data memory, and it requires 709,878 clocks, i.e. about 3.5 ms, for processing a frame of 20 ms speech signal.

• 한국음향학회지
• /
• 제20권8호
• /
• pp.19-23
• /
• 2001

고속 저전력의 DSP (Programmable Digital Signal Processor)가 개발됨에 따라 이동통신 분야에서 시스템 및 단말기 등이 DSP를 사용하여 구현되고 있다. 본 논문에서는 DSP를 사용한 AMR (Adaptive Multi-rate) 음성부호화기의 멀티 채널 실시간 구현에 관하여 논한다. AMR 음성부호화 알고리즘을 250 MHz로 동작하는 32비트 정수형 DSP 칩인 TMS320C6202를 사용하여 구현하였다. 실시간 동작을 위하여 cross compile, 선형 어셈블리 최적화, TMS320C62xx 어셈블리 최적화 작업을 수행하였다. AMR 음성부호화기에 음성 데이터 입출력 기능 및 외부 CPU와의 통신기능을 포함하였다. DSP EVM 보드를 사용하여 AMR 음성부호화기를 개발하였고, ETRI에서 개발중인 비동기 IMT-2000 시스템 상에서 동작 및 기능을 검증하였다.

• 한국컴퓨터정보학회논문지
• /
• 제21권12호
• /
• pp.131-137
• /
• 2016

In this paper, the peak finding algorithms corresponding to the Autocorrelation Function (ACF), which are widely exploited for detecting the pitch of voiced signal, are proposed. According to various researchers, it is well known fact that the estimation of fundamental frequency (F0) in speech signal is not only very important task but quite difficult mission. The proposed algorithms, presented in this paper, are implemented by using many characteristics - such as monotonic increasing function - of ACF function. Thus, the proposed algorithms may be able to estimate both reliable and correct the fundamental frequency as long as the autocorrelation function of speech signal is accurate. Since the proposed algorithms may reduce the computational complexity it can be applied to the real-time processing. The speech data, is composed of Korean emotion expressed words, is used for evaluation of their performance. The pitches are measured to compare the performance of proposed algorithms.

• 전자공학회논문지CI
• /
• 제41권2호
• /
• pp.1-8
• /
• 2004

자동차 텔레매틱스 용 음성 HMI(Human Machine Interface) 기술은 차량 내 음성정보기술 활용을 위하여 차량 잡음환경에 강인한 내장형 음성 기술을 통합한 음성 HMI 기반 텔레매틱스 용 DSP 시스템의 개발을 포함한다. 개발된 내장형 음성 인식엔진을 바탕으로 통합 시험을 위한 자동차 텔레매틱스 용 DSP 시스템 구현 개발을 수행하는 본 논문은 자동차용 음성 HMI의 요소 기술을 통합하는 기술 개발로 자동차용 음성 HMI 기술 개발에 중심이 되는 연구이다.

• 대한의용생체공학회:의공학회지
• /
• 제11권1호
• /
• pp.75-82
• /
• 1990

The purpose of this study is to help the hearing Impaired's speech training with a visible speech processing system. In brief, this system converts the features of speech signals into graphics on monitor, and adjusts the features of hearing impaired to normal ones. There are formant and pitch in the features used for this system. They are extracted using the digital signal processing such as linear predictive method or AMDF(Average Magnitude Difference Function). In order to effectively train for the hearing impaired's abnormal speech, easilly visible feature has been being studied.

• 한국정보통신학회논문지
• /
• 제11권5호
• /
• pp.961-968
• /
• 2007

오늘날, 우리는 어디엔가 엔제나 무전기 통신 장치를 사용할수 있다. 때때로, 우리는 음향잡음환경에서 장치를 사용하였다. 그 음향잡음은 통신장치에서 많은 문제를 만들었다. 음향잡음환경에서는, 말은 음성신호와 잡음신호 양쪽에 신호를 포함하고, 받았기 때문에 깨끗한 정보를 받기위해 보낼수가 없었다. 디지털필터는 바라는 신호를 얻기 위해 옮기는 잡음으로서 유용하였다. 방법의 하나는 자동적으로 맞추는 필터 파라미터로서 적응 잡음 망상조직으로 적응디지털필터를 사용하는 것이다. 본 논문은 두 적응필터 방법에 의하여 현실에서 음향잡음으로서 명료도 알고리즘의 번지라고 할 수가 있다. 하나는 두 입력 채널과 함께 적응잡음 망상조직이라 할 수 있고, 또 다른 것은 하나 입력 채널과 함께 스펙트럼 빼기필터이다. 이 실험의 결과는 제안된 필터로부터 스펙트럼 진폭필터는 움직이지 않는 잡음은 효력이 있는 동안 움직이는 것을 줄이기 위해 사용되어지는 것은 적응잡음망상조직으로 보여준다.

• 센서학회지
• /
• 제28권5호
• /
• pp.299-304
• /
• 2019

Recently, the development of rehabilitation medical technology has resulted in an increased interest in speech therapy equipment. In particular, research on articulation therapy for communication disorders is being actively conducted. Existing methods for the diagnosis and treatment of speech disorders have many limitations, such as traditional tactile perception tests and methods based on empirical judgment of speech therapists. Moreover, the position and tension of the tongue are key factors of speech disorders with regards to articulation. This is a very important factor in the distinction of Korean characters such as lax, fortis, and aspirated consonants. In this study, we proposed a Korean electropalatography (EPG) system to easily measure and monitor the position and tension of the tongue in articulation treatment and diagnosis. In the proposed EPG system, a sensor was fabricated using an AgCl electrode and biocompatible silicon. Furthermore, the measured signal was analyzed by implementing the bio-signal processing module and monitoring program. In particular, the bio-signal was measured by inserting it into the palatal from an experimental control group. As a result, it was confirmed that it could be applied to clinical treatment in speech therapy.