• 한국정보통신학회논문지
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• 제10권4호
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• pp.739-743
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• 2006

본 논문은 신경회로망을 이용한 화자적응 음성인식 제어시스템을 개발하였다. 화자종속시스템은 단일 화자의 음성만 등록하여 이용하므로 여러 화자의 음성을 인식하는 데는 문제가 있고, 화자독립시스템은 여러 화자를 인식한다. 본 연구 개발에서는 화자적응시스템을 구현하여 화자종속형의 단점을 보완하여 화자 독립과 화자 종속을 혼합하여 사용 할 수 있는 기능으로 화자 적용방법으로 구현하였고, 화자인증(Speaker Verification)도 가능하도록 프로그램 하였다.

• 제어로봇시스템학회논문지
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• 제16권4호
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• pp.335-338
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• 2010

A new method that improves the performance of large scale voice dialling system is presented using speaker adaptation. Since SI (Speaker Independent) based speech recognition system with phoneme HMM uses only the phoneme string of the input sentence, the storage space could be reduced greatly. However, the performance of the system is worse than that of the speaker dependent system due to the mismatch between the input utterance and the SI models. A new method that estimates the phonetic string and adaptation vectors iteratively is presented to reduce the mismatch between the training utterances and a set of SI models using speaker adaptation techniques. For speaker adaptation the stochastic matching methods are used to estimate the adaptation vectors. The experiments performed over actual telephone line shows that proposed method shows better performance as compared to the conventional method. with the SI phonetic recognizer.

• 한국위성정보통신학회논문지
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• 제10권3호
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• pp.22-25
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• 2015

본 논문에서는 PA스피커 시설물 부착형 LED패치의 음원감성 연계형 컬러코드 제어 기술에 관한 연구를 진행하였다. PA스피커 음원에 따라 LED패치의 컬러코드를 제어할 수 있는 기술을 제시하였으며, 이를 위한 PA스피커 시설물 부착형 LED 패치를 개발하였다. PA스피커 시설물 부착형 LED패치의 컬러코드 제어 기술은 PA스피커로부터 음원 감지 유무를 확인하고, 음원이 감지된 경우 아날로그 신호(음원)을 디지털 신호로 변환하여 메인컨트롤러에 전달하여 LED패치의 컬러코드 색상 및 패턴을 제어할 수 있도록 구현하였다. 본 논문에서는 PA스피커 시설물 부착형 LED패치의 음원감성 연계형 컬러코드 제어 시스템을 구축하고, 이를 통하여 PA스피커의 음원에 따라 효과적으로 컬러코드를 제어할 수 있는 PA스피커 시설물 부착형 LED패치 컬러코드 제어 실험을 진행하였으며, LED패치의 컬러코드 제어 유무를 통한 제안기술의 유용성을 보였다. PA스피커 시설물 부착형 LED패치의 음원감성과 연계된 컬러코드 제어 기술은 향후 다양한 분야에 연구 방향을 제시하고, 응용사례로 널리 활용될 수 있을 것으로 예상된다.

• 소음진동
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• 제4권2호
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• pp.187-198
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• 1994

This study is to describe the effects of the duct termination conditions conditions upon the active noise attenuation system. The adaptive filtering algorithm using FIR filter is implemented for duct noise attenuation. To avoid the instability caused by the acoustic feedback, two methods are considered. One is to use a compensating FIR filter. The other is to utilize uni-directional detecting microphone and uni-directional control speaker. Experimental results show that the reflections of sound from duct terminations greatly reduce the performance of ANC system. The directionality of detecting microphone and control speaker is a major factor to decide ANC performance. When there are some reflections from both duct terminations, the noise attenuation using finite FIR filter is not enough to model the duct plant. Especially, the reflection from the upstream termination reduces the noise attenuation in the frequencies related to the distance between control speaker and upstream termination. The performance of the noise attenuation is found to be largely enhanced by using uni-directional coupler, both on detecting microphone and control speaker, even if the duct system has an arbitrary termination conditions.

• 한국음향학회지
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• 제40권3호
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• pp.247-253
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• 2021

소리에 방향성을 부여하여 특정 영역에서만 소리를 들을 수 있도록 활용되는 초음파 스피커는 일반 스피커와 비교하여 음질 및 비용적인 이슈에서 다양한 개선 연구가 지속적으로 이루어지고 있다. 본 논문에서는 초음파 스피커의 센서 특성상 500 Hz 미만 저음 구현이 어려운 점을 감안하여 500 Hz 대의 소리를 보완할 수 있도록 일반 스피커와 동시 사용이 가능한 DSP 기반의 하이브리드형 초음파 스피커를 제안한다. 일반 스피커와 초음파 스피커의 단순 연결로 각각의 분리 처리 및 송출하는 시스템은 초음파 재생성 처리 시간 차에 따른 음질저하뿐만 아니라 일반 음원과 초음파 음원이 2개의 앰프로 구동되어 높은 비용 이슈가 있으며 제반 제어적인 측면에서도 어려움이 있다. 이러한 점을 개선하고자 제안한 DSP 기반의 앰프에서 Dynamic Range Control(DRC) 및 Equalizer(EQ)의 기존 코덱 기능은 물론, 초음파 음원으로의 재생성, 일반/초음파 음원을 동기화함으로써 동시 재생이 가능한 하이브리드형 초음파 스피커를 개발하였다.

• 대한음성학회지:말소리
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• 제49호
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• pp.145-158
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• 2004

This paper is aiming at implementation of real-time speaker verification system using DSP board. Dialog/4, which is based on microprocessor and DSP processor, is selected to easily control telephone signals and to process audio/voice signals. Speaker verification system performs signal processing and feature extraction after receiving voice and its ID. Then through computing the likelihood ratio of claimed speaker model to the background model, it makes real-time decision on acceptance or rejection. For the verification experiments, total 15 speaker models and 6 background models are adopted. The experimental results show that verification accuracy rates are 99.5% for using telephone speech-based speaker models.

• ETRI Journal
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• 제40권5호
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• pp.643-652
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• 2018

Dysarthria is a degenerative disorder of the central nervous system that affects the control of articulation and pitch; therefore, it affects the uniqueness of sound produced by the speaker. Hence, dysarthric speaker recognition is a challenging task. In this paper, a feature-extraction method based on deep belief networks is presented for the task of identifying a speaker suffering from dysarthria. The effectiveness of the proposed method is demonstrated and compared with well-known Mel-frequency cepstral coefficient features. For classification purposes, the use of a multi-layer perceptron neural network is proposed with two structures. Our evaluations using the universal access speech database produced promising results and outperformed other baseline methods. In addition, speaker identification under both text-dependent and text-independent conditions are explored. The highest accuracy achieved using the proposed system is 97.3%.

• 한국항해항만학회지
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• 제35권7호
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• pp.551-556
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• 2011

In this paper new enhanced post-process predicting the speaker's intention was suggested to implement the real-time control module for ship's autopilot using speech recognition algorithm. The parameter was developed to predict the likeliest wheel order based on the previous order and expected to increase the recognition rate more than pre-recognition process depending on the universal speech recognition algorithms. The values of parameter were assessed by five certified deck officers being good at conning vessel. And the entire wheel order recognition process were programmed to TMS320C5416 DSP so that the system could recognize the speaker's orders and control the autopilot in real-time. We conducted some experiments to verify the usefulness of suggested module. As a result, we have confirmed that the post-recognition process module could make good enough accuracy in recognition capabilities to realize the autopilot being operated by the speech recognition system.

• 대한임베디드공학회논문지
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• 제16권3호
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• pp.89-97
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• 2021

Voice interface can not only make daily life more convenient through artificial intelligence speakers but also improve the working environment of the factory. This paper presents a voice-assisted work management system that supports both speech and speaker recognition. This system is able to provide machine control and authorized worker authentication by voice at the same time. We applied two speech recognition methods, Google's Speech application programming interface (API) service, and DeepSpeech speech-to-text engine. For worker identification, the SincNet architecture for speaker recognition was adopted. We implemented a prototype of the work management system that provides voice control with 26 commands and identifies 100 workers by voice. Worker identification using our model was almost perfect, and the command recognition accuracy was 97.0% in Google API after post- processing and 92.0% in our DeepSpeech model.

• 산업기술연구
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• 제29권A호
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• pp.95-100
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• 2009

In this paper, we implemented a speaker-independent speech recognizer using the TMS320F28335 DSP which is optimized for control applications. For this implementation, we used a small-sized commercial DSP module and developed a peripheral board including a codec, signal conditioning circuits and I/O interfaces. The speech signal digitized by the TLV320AIC23 codec is analyzed based on MFCC feature extraction methed and recognized using the continuous-density HMM. Thanks to the internal SRAM and flash memory on the TMS320F28335 DSP, we did not need any external memory devices. The internal flash memory contains ADPCM data for voice response as well as HMM data. Since the TMS320F28335 DSP is optimized for control applications, the recognizer may play a good role in the voice-activated control areas in aspect that it can integrate speech recognition capability and inherent control functions into the single DSP.