• Journal of Sensor Science and Technology
• /
• v.15 no.4
• /
• pp.223-230
• /
• 2006

An FIR filter for digital voice signal processing has been designed and implemented using a microcontroller in implantable middle ear hearing devices (IMEHDs). The designed digital voice signal processing filter which has fast and accurate filtering operation and controllable filter characteristics has been implemented using a hardware multiplier and a direct memory access (DMA) in the low power microcontroller, MSP430F169. It has been confirmed that each of the implemented 6-orders Remez FIR filters with 1 channel and 2 channels can be applied to the voice signal processing module of IMEHDs based on the evaluation results of the filtering performance experiment.

• Proceedings of the Korea Institute of Convergence Signal Processing
• /
• 2005.11a
• /
• pp.290-293
• /
• 2005

음성 인식 기술이 발달함에 따라 음성 인식 기술을 이용한 응용의 개발이 중요한 문제로 떠오르고 있다. VoiceXML은 전화기를 통한 음성 인터페이스를 위한 XML 언어로서 손쉬운 방법으로서 음성 인터페이스를 설계, 구현할 수 있도록 만들어진 언어이다. 본 논문에서는 이를 이용해 전화를 통하여 음성으로 자동차 정보 안내 시스템을 사용할 수 있는 사용자 인터페이스를 구현한다. 구현된 시스템 및 서비스는 VoiceXML의 장점을 활용하여 원거리에서 편리하게 사용자가 자동차의 정보를 안내받고 제어할 수 있는 인터페이스 자체보다는 음성 인터페이스의 설계 및 구현에 중점을 두었다. 10인의 피실험자가 각 10회씩 총 100회를 실험한 결과 99.3%의 인식률을 보였다. 추후 차세대 자동차 텔레메틱스 서비스와 연동하면 구현되어진 시스템의 활용이 증대될 것이라 판단된다.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.40 no.11
• /
• pp.42-50
• /
• 2003

In this paper we propose a method of evaluating the performance of a Softswitch that provides call control to voice services in NGN (next generation network). First, we describe the architecture for voice services in NGN and anatomize the call control processes such as call initiation, call re-initiation and call release of a voice connection. kiter that we propose a method of estimating appropriate server capacity of the Softswitch using approximate queuing model. Via numerical experiments we illustrate the implication of the work

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.2
• /
• pp.173-181
• /
• 2002

Generally the one of serious problems in Voice Activity Detection (VAD) is speech region detection in noise environment. Therefore, this paper propose the new method using energy, lsp varation. As a result of processing time and speech quality of the proposed algorithm, the processing time is reduced due to the accurate detection of inactive period, and there is almot no difference in the subjective quality test. As a result of bit rate, proposed algorithm measures the number of VAD=1 and the result shows predominant reduction of bit rate as SNR of noisy speech is low (about 5∼10 dB).

• The Transactions of the Korea Information Processing Society
• /
• v.3 no.6
• /
• pp.1506-1514
• /
• 1996

This paper proposes enhanced features for the Universal Telecommunications (UPT), voice authentication and voice synthesis, using the specialized resources functions in the Intelligent peripheral(IP). The proposed voice authentication is able ti provide simple and user-friendly security mechanism and to prevent unauthorized users from fraudulently using the UPT number. Also, traditional UPT service deliveries only fixed message to the UPT user, but the proposed UPT service can support flexible message transfer by use of the voice synthesis.

• Journal of the Korean Society of Industry Convergence
• /
• v.21 no.1
• /
• pp.17-27
• /
• 2018

We propose a new approach to control a biped robot motion based on iterative learning of voice command for the implementation of smart factory. The real-time processing of speech signal is very important for high-speed and precise automatic voice recognition technology. Recently, voice recognition is being used for intelligent robot control, artificial life, wireless communication and IoT application. In order to extract valuable information from the speech signal, make decisions on the process, and obtain results, the data needs to be manipulated and analyzed. Basic method used for extracting the features of the voice signal is to find the Mel frequency cepstral coefficients. Mel-frequency cepstral coefficients are the coefficients that collectively represent the short-term power spectrum of a sound, based on a linear cosine transform of a log power spectrum on a nonlinear mel scale of frequency. The reliability of voice command to control of the biped robot's motion is illustrated by computer simulation and experiment for biped walking robot with 24 joint.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.13 no.2
• /
• pp.171-181
• /
• 1988

In this study, a packet voice/data terminal(PVDT) that services both voice and data in the packet-switched network is implemented. The software structure of the PVDT is designed according to the OSI 7 layer architecture. The discrimination of voice and data is made in the link layer. Voice packets have priority over data packets in order to minimize the transmission delay, and are serviced by a simple protocol so that the overhead arising form the retransmission of packets may be minimized. The hardware structure of the PVDT is divided into five modules; a master control module, a speech proessing module, a speech activity detection module, a telephone interface module, and an input/output interface module. In addition to the hardware implementation, the optimal reconstruction delay of voice packets to reduce the influence of delay variance is analyzed.

• Journal of the Institute of Convergence Signal Processing
• /
• v.9 no.3
• /
• pp.208-212
• /
• 2008

This paper proposes a model of a voice communication using bluetooth and a mobile, and implements a hands-free for this model. A Proposed model uses bluetooth for a voice network, and is possible to share a voice with a great number of people using preemptive algorithm proposed in this paper. There are two types of models. One is a model only using a hands-free, the other is a model using a mobile. Second model uses a scatternet and a call forwarding service for a call on a voice communication. In case of using a scatternet, scatternet is composed of a piconet of a voice communication and a piconet of a call. In case of using a call forwarding service, the mobiles share information of each others before formation of a network.

• Phonetics and Speech Sciences
• /
• v.4 no.2
• /
• pp.127-134
• /
• 2012

The purpose of this study is to present evidence for a particular processing approach based on the language-specific characteristics of Korean. To compare individuals' sentence-comprehension abilities, this study measured the accuracy and reaction times (RT) of 12 aphasic patients (AP) and 12 normal controls (NC) during a sentence-picture matching task. Four versions of a sentence were constructed with the two types of voice (active/passive) and two types of word order (agent-first/patient-first). To examine the effects of increased processing demand, picture stimuli were manipulated in such a way that they appeared immediately after the sentence was presented. As expected, the AP group showed higher error rates and longer RT for all conditions than the NC group. Furthermore, Korean speakers with aphasia performed above a chance level in sentence comprehension, even with passive sentences. Aphasics understood sentences more quickly and accurately when they were given in the active voice and with agent-first order. The patterns of the NC group were similar. These results confirm that Korean adults with aphasia do not completely lose their knowledge of sentence comprehension. When the processing demand was increased by delaying the picture stimulus onset, the effect of increased processing demands on RT was more pronounced in the AP than in the NC group. These findings fit well with the idea that the computational system for interpreting sentences is intact in aphasics, but its ability is compromised when processing demands increase.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.10 no.8
• /
• pp.901-906
• /
• 2015

We presented a method for Vad(Voice Activity Detection) using Bi-level HMM. Conventional methods need to do an additional post processing or set rule-based delayed frames. To cope with the problem, we applied to VAD a Bi-level HMM that has an inserted state layer into a typical HMM. And we used posterior ratio of voice states to detect voice period. Considering MFCCs(: Mel-Frequency Cepstral Coefficients) as observation vectors, we performed some experiments with voice data of different SNRs and achieved satisfactory results compared with well-known methods.