• Journal of the Institute of Convergence Signal Processing
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• v.10 no.3
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• pp.170-175
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• 2009

In this paper, we propose real time automatic speaker verification (ASV) system to protect personal information on personal digital assistant (PDA) device. Recently, the capacity of PDA has extended and been popular, especially for mobile environment such as mobile commerce (M-commerce). However, there still exist lots of difficulties for practical application of ASV utility to PDA device because it requires too much computational complexity. To solve this problem, we apply the method to relieve the computational burden by performing the preprocessing such as spectral subtraction and speech detection during the speech utterance. Also by applying the hidden Markov model (HMM) optimal state alignment and the sequential probability ratio test (SPRT), we can get much faster processing results. The whole system implementation is simple and compact enough to fit well with PDA device's limited memory and low CPU speed.

• Proceedings of the KOSOMBE Conference
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• v.1997 no.11
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• pp.63-66
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• 1997

With the advent of high speed digital signal processing chips, new digital techniques have been introduced to the hearing instrument. This advanced hearing instrument circuitry has led to the need or and the development of new fitting approach. A number of different fitting approaches have been developed over the past few years, yet there has been little agreement on which approach is the "best" or most appropriate to use. However, when we develop not only new hearing aid, but also its fitting method, the intensive subject-based clinical tests are necessarily accompanied. In this paper, we present an objective method to evaluate and predict the performance of hearing aids without the help of such subject-based tests. In the hearing impairment simulation (HIS) algorithm, a sensorineural hearing impairment model is established from auditory test data of the impaired subject being simulated. Also, in the hearing impairment simulation system the abnormal loudness relationships created by recruitment was transposed to the normal dynamic span of hearing. The nonlinear behavior of the loudness recruitment is defined using hearing loss unctions generated from the measurements. The recruitment simulation is validated by an experiment with two impaired listeners, who compared processed speech in the normal ear with unprocessed speech in the impaired ear. To assess the performance, the HIS algorithm was implemented in real-time using a floating-point DSP.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.28 no.3
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• pp.133-141
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• 2005

The previous research has been mainly focused on individual technology of voice recognition, voice synthesis, translation, and bone transmission technical. Recently, commercial models have been produced using aforementioned technologies. In this research, a new automated translation support system concept has been proposed by combining established technology of bone transmission and wireless system. The proposed system has following three major components. First, the hybrid system consist of headset, bone transmission and other technologies will recognize user's voice. Second, computer recognized voice (using small server attached to the user) of the user will be converted into digital signal. Then it will be translated into other user's language by translation algorithm. Third, the translated language will be wirelessly transmitted to the other party. The transmitted signal will be converted into voice in the other party's computer using the hybrid system. This hybrid system will transmit the clear message regardless of the noise level in the environment or user's hearing ability. By using the network technology, communication between users can also be clearly transmitted despite the distance.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.200-206
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• 2006

This paper presents a preprocessing method to modify the input audio signals of a speech coder to obtain the finally enhanced signals at the decoder. For the purpose, we introduce the noise suppression (NS) scheme and the adaptive gain control (AGC) where an audio input and its coding error are considered as a noisy signal and a noise, respectively. The coding error is suppressed from the input and then the suppressed input is level aligned to the original input by the following AGC operation. Consequently, this preprocessing method makes the spectral energy of the music input redistributed all over the spectral domain so that the preprocessed music can be coded more effectively by the following coder. As an artifact, this procedure needs an additional encoding pass to calculate the coding error. However, it provides a generalized formulation applicable to a lot of existing speech coders. By preference listening tests, it was indicated that the proposed approach produces significant enhancements in the perceived music qualities.

• Journal of Biomedical Engineering Research
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• v.34 no.1
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

• KIISE Transactions on Computing Practices
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• v.21 no.8
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• pp.531-536
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• 2015

The speech recognition is an active research area in the human computer interface (HCI). The objective of this study is to control digital devices with voices. In addition, the mouse is used as a computer peripheral tool which is widely used and provided in graphical user interface (GUI) computing environments. In this paper, we propose a method of controlling the mouse with the real-time speech recognition function of a smartphone. The processing steps include extracting the core voice signal after receiving a proper length voice input with real time, to perform the quantization by using the learned code book after feature extracting with mel frequency cepstral coefficient (MFCC), and to finally recognize the corresponding vowel using hidden markov model (HMM). In addition a virtual mouse is operated by mapping each vowel to the mouse command. Finally, we show the various mouse operations on the desktop PC display with the implemented smartphone application.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.768-771
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• 2004

In this paper, we propose a content-based audio genre classification algorithm that automatically classifies the query audio into five genres such as Classic, Hiphop, Jazz, Rock, Speech using digital signal processing approach. From the 20 second query audio file, 54 dimensional feature vectors, including Spectral Centroid, Rolloff, Flux, LPC, MFCC, is extracted from each query audio. For the classification algorithm, k-NN, Gaussian, GMM classifier is used. In order to choose optimum features from the 54 dimension feature vectors, SFS (Sequential Forward Selection) method is applied to draw 10 dimension optimum features and these are used for the genre classification algorithm. From the experimental result, we verify the superior performance of the SFS method that provides near $90{\%}$ success rate for the genre classification which means $10{\%}$-$20{\%}$ improvements over the previous methods

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.131-135
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• 2000

본 논문에서는 고성능 고정 소수점 DSP (Digital Signal Processor) 코어인 Vincent6 코어 [1]를 이용하여 ITU-T C.728 음성 부호화기를 실시간으로 구현하였다 G.728 은 16 kb/s전송률의 ITU-T표준 음성 부호화기이며, 입력신호는 8 kHz로 샘플링되며 샘플 당 16 bit 로 양자화된 PCM 신호이다. G.728 은 LD-CELP(Low Delay Code Excited Linear Prediction)라고도 하며, 알고리 듬 delay는 0.625ms 이다. Vincent6 DSP core 는 VLIW (Very-Long Instruction Word) 특성을 가지므로 다중 명령 (multiple instruction)을 수행할 수 있다 이를 위해서 G.728 annex G를 이용하여 고정 소숫점 연산으로 코드를 작성한 후, 이를 vincent6 어셈블리 코드로 구현하였다. 최종적으로 구현된 코드는 ITU-T 의 test vector 에 대 해 bit exact 한 결과를 보이며 34 MCPS (Million Cycles Per Second)의 계산량을 가지며 사용 메모리크기는 데이터 메모리가 약 9KByte, 프로그램 메모리가 약 57 KByte 이다.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.3
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• pp.1-9
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• 1993

This paper discusses issues on the disign and implementation of real-time automatic speaker verification system, as well as the performance analysis of the implemented system. The system employs TI's TMS320C25 digital signal processor TMS320C25 and high speed SRAMs. The system is designed to be used stand-alone as well as via hand-shaking with IBM-PC. The speech parameters used for speaker verification are PARCOR and LPC-cepstrum coefficients, and the employed decision logics are those based on the generalized weighted distance comcept. The implemented system showed the performance of 5.3% error rate for the PARCOR coefficient, and 4.7% error rate for the LPG-cepstrum coefficient.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.07a
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• pp.254-256
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• 2012

본 논문에서는 MPEG-D 오디오 서브그룹에서 진행 중인 Unified Speech and Audio Coding (USAC) 표준의 Linear Prediction Domain (LPD) 복호화기 모듈을 고정소수점 알고리즘으로 제안한다. USAC 부호화기는 두 개의 최신 음성-오디오 부호화기가 융합된 형태로, 음성 및 오디오 신호에 대하여 우수한 성능을 갖는 부호화기이다. USAC의 표준 완료와 본격적인 서비스화에 앞서서 USAC LPD 복호화기의 구조적인 특성을 분석하고, Digital Signal Processor (DSP)구현을 위한 LPD 복호화기의 고정소수점 알고리즘을 구축하는 동시에 모듈의 복잡도를 측정하고자 한다. 또한 고정소수점 알고리즘으로 구현된 LPD 복호화기와 기존의 부동소수점 복호화기의 성능을 비교하고, LPD 복호화기의 두 가지 부호화 모드에 따른 복잡도 이슈를 다루도록 한다.