• Journal of the Institute of Electronics Engineers of Korea SD
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• v.41 no.5
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• pp.91-97
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• 2004

A digital over-sampling noise-shaping coder to achieve the processing accuracy for the audio signal bandwidth is designed. In order to implement an optimized design of the noise-shaping coder as a form of U (intellectual property), circuit design techniques that optimize the multiplication and the ROM architectures are proposed with emphasis on the low-voltage operation under 2.0 V and the minimization of the hardware resources. In the design and verification methodology, the overall architectures and the internal bit width have been determined through behavioral simulations. The overall performances including timing margin have been estimated through transistor-level simulations. Furthermore, the test results of the implemented chip using a 0.35-${\mu}{\textrm}{m}$ standard CMOS process proposed the validity of the proposed circuits and the design methodology.

• The Journal of the Acoustical Society of Korea
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• v.15 no.5
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• pp.90-98
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• 1996

An auditory model is proposed for robust speech recognition in noisy environments. The model consists of cochlear bandpass filters and nonlinear stages, and represents frequency and intensity information efficiently even in noisy environments. Frequency information of the signal is obtained by zero-crossing intervals, and intensity information is also incorporated by peak detectors and saturating nonlinearities. Also, the robustness of the zero-crossings in estimating frequency is verified by the developed analytic relationship of the variance of the level-crossing interval perturbations as a function of the crossing level values. The proposed auditory model is computationally efficient and free from many unknown parameters compared with other auditory models. Speaker-independent speech recognition experiments demonstrate the robustness of the proposed method.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.5
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• pp.59-63
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• 2011

We present a lattice transversal joint (LTJ) adaptive filter with fixed reflection coefficients to achieve fast convergence with low complexity. The reflection coefficients of the filter are given by the statistics of speech signals, and the proposed order of the lattice predictor is one. Experimental results confirm that as compared to the adaptive transversal filter, the proposed adaptive filter achieves fast convergence with a negligible increase in complexity. The proposed adaptive filter converges around six times faster than the adaptive transversal filter in case of the band-limited voiced signal from the ITU-T G.168 standard.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.719-722
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• 2001

최근 정지영상, 동영상, 음성 등의 멀티미디어 컨텐트가 디지털화 되고 네트워크가 발달함에 따라 영상을 포함한 멀티미디어 데이터의 접근이 용이해 졌다. 이러한 데이터의 불법적 사용과 인위적인 조작으로부터 소유권과 저작권을 효율적으로 보호하기 위한 워터마킹 기술이 많이 연구되고 있다. 일반적으로 정지 영상 및 음성에 대한 워터마킹 기술은 많이 연구가 되었지만 이러한 방법을 동영상에 그대로 적용하기에는 실시간 처리에 적용하기가 힘들다는 큰 문제점이 있다. 따라서 본 논문에서는 비디오 신호에서의 빠른 처리과정과 실시간으로 워터마크를 삽입하고, 원 영상 없이 워터마크를 추출 할 수 있는 새로운 방법(Blind Watermarking)을 제안하고자 한다. 제안한 방법은 대역확산을 근거로 하여 워터마크 은닉 과정에서 치환(Permutation) 과정과 LSB 부호화 방법을 이용하여 비디오 시퀀스의 모든 I-프레임에 은닉한다. 복원과정은 모든 I-프레임에서 LSB 복호화와 역 치환 과정을 거쳐 본래의 저작권 정보를 추출한다. 제안한 방법을 여러 가지 동영상 비디오에 적용해 본 결과 기존의 워터마킹 방법보다 효율적이고 시각적 손상이 없었으며 빠른 실시간 처리가 가능함을 볼 수 있었다.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.4
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• pp.24-30
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• 1984

Group delay distortion (phase distortion) is a characteristic which is of no account from a standpoint of voice transmission. But this distortion becomes the major source of distortion in wave form transmission such as data, FAX and others over the public switching telephone network (voice band transmission) so that it must be drastically studied. This paper makes analysis of group deray distortion of the telephone network, describs experimental and measuring results and refers also to the improvement of distortion for the purpose of opening the public switching telephone network to data transmission.

• Journal of the Korean Institute of Telematics and Electronics
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• v.20 no.3
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• pp.29-32
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• 1983

This paper is an experimental study on the generation principles of digital frequency using ROM-table look-up method and the design of the MF digital Sender used in signalling systems between ESS. After construction of MF digital Sender, through experiment, we concluded that this system well suit for CCITT (International Telegraph and Telephone Consultative Committee) recommendation and this basic principle can be applied to the signalling method using frequency Within voice-band. Also it can be applied to R2 MFC signalling equipment which is used between electronic switching systems (ESS) signalling system.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.589-598
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• 2009

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music content MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved WD3 at the 88th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC ACELP and TCX) for low frequency regions, SBR for high frequency regions and the MPEG Surround tool for stereo information. USAC can provide consistent sound quality for both speech and music content and can be applied to various applications such as multi-media download to mobile device Digital radio Mobile TV and audio books.

• Journal of Digital Convergence
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• v.12 no.6
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• pp.271-276
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• 2014

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and unvoiced consonants in a frame. In this paper, I propose a method of 8kbps Multi-Pulse Speech Coding(FBD-MPC: Frequency Band Division MPC) by using TSIUVC(Transition Segment Including Unvoiced Consonant) searching, extraction and approximation-synthesis method in a frequency domain. I evaluate the 8kbps MPC and FBD-MPC. As a result, SNRseg of FBD-MPC was improved 0.5dB for female voice and 0.2dB for male voice respectively. Compared to the MPC, SNRseg of FBD-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• The Journal of the Acoustical Society of Korea
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• v.15 no.4
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• pp.83-92
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• 1996

This paper describes Lombard effect compensation and noise suppression so as to reduce speech recognition error in noisy environments. Lombard effect is represented by the variation of spectral envelope of energy normalized word and the variation of overall vocal intensity. The variation of spectral envelope can be compensated by linear transformation in cepstral domain. The variation of vocal intensity is canceled by spectral magnitude normalization. Spectral subtraction is use to suppress noise contamination, and band-pass filtering is used to emphasize dynamic features. To understand Lombard effect and verify the effectiveness of the proposed method, speech data are collected in simulated noisy environments. Recognition experiments were conducted with contamination by noise from automobile cabins, an exhibition hall, telephone booths in down town, crowded streets, and computer rooms. From the experiments, the effectiveness of the proposed method has been confirmed.

• ETRI Journal
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• v.9 no.3
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• pp.74-85
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• 1987

In this paper, a new kind of dual tone multifrequency receiver(DTMFR) is proposed, in which detection is made in terms of the partial correlation(PARCOR) coefficients for efficient DTMF signal detection against both temporary decoding errors by crosstalks possibly coming from adjoint telephone lines during transmission and so-called digit simulation by background voice or noise signals during interdigit period. A simulation study on the behaviour of PARCOR coefficients for tone signals and non-signals has been performed in order to provide the rationale on the feasibility of the proposed DTMFR algorithm. Based upon simulation results, a more refined detection strategy as an example is presented and explained together with the corresponding decision logic.