• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.19 no.5
• /
• pp.912-919
• /
• 1994

A new error control line code based on an extended Hamming code is proposed and its performance is analyzed in this paper. The proposed code is capable of single error correction and double error detection since its minimum Hamming distance is 4. In addition, the error detection capability can be oncreased due to the redundancy bit used for line coding. As a result, the proposed code shows lower code rate, but better spectral characteristics in low frequency region and lower residual bit error rate than the conventional error correction line code using Hamming (7, 4) code.

• The Journal of the Acoustical Society of Korea
• /
• v.3 no.2
• /
• pp.69-90
• /
• 1984

본 논문에서는 잡음 spectrum 변형 기능을 가진 APCM, ADPCM 및 ADM 음성 부호기의 성능 에 관해서 연구하였다. 잡은 SPECTRUM 변형방식은 두가지를 고려할 수 있는데, APCM과 ADPCM에 서는 C-massage weighting 된 양자화 잡음을 최소화하는 noise feedback filter를 이용하는 방법을 채택 하고, ADM에서는 in-band의 잡음의 일부를 신호대역의 밖으로 옮기는 방법을 사용하였다. APCM 과 ADPCM 부호기의 성능을 측정하는데는 주파수가 weighting이 된 신호대 잡음비와 segment된 FWSQNR를 사용하였다. 실제음성을 사용한 simulation 결과에 의하면 잡음 spectrum 변형기능을 가진 부호기가 없는 것보다 0.5 내지 3dB 가량 좋은 것으로 나타났다. 이러한 개선은 양적으로 비교적 적은 것이 사실이지만 실제로 음성을 들어보면 음질이 현저히 좋아짐을 알 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.22 no.4
• /
• pp.737-747
• /
• 1997

performance of the cellular CDMA overlay system is analyzed which shares the same band with existing microwave narrowband system to enhance the spectral efficiency. To suppress the interfreence from narrowband system, we used a linear predicition filter that adopts the adaptive least mean square algorithm. Alalyzing the performance in the Personal Communication Services channel, characterized as a multipath Rician fadng channels, we considered the power control to solve the near-far problem, and-off and multipath diversity. We also considered interleaving and channel coding to improve BER performance of the CDMA system.

• The Journal of the Acoustical Society of Korea
• /
• v.25 no.1
• /
• pp.21-29
• /
• 2006

In this Paper a split bandwidth wideband speech coder and its highband coding method are Proposed. The coder uses a split-band approach. where the wideband input speech signal is split into two equal frequency bands from 0-4kHz and 4-8kHz. The lowband and the highband are coded respectively by the 11.8kb/s G.729 Annex E and the proposed coding method. After the LPC analysis, the highband is divided by two modes according to the properties of signals. In stationary mode. the highband signals are compressed by the mixture excitation model; CELP algorithm and W (Matching Pursuit) algorithm. The others are coded by the only CELP algorithm. We compare the performance of the new wideband speech coder with that of G.722 48kbps SB-ADPCM and G.722.2 12.85kbps in a subjective method. The simulation results show that the Performance of the proposed wideband speech coder has better than that of 48kbps G.722 and no better than that of 12.85kbps G.722.2.

• Journal of the Korea Society of Computer and Information
• /
• v.18 no.7
• /
• pp.23-35
• /
• 2013

This paper proposes an effective technique that is used to automatically extract feature vectors from vibration signals for fault classification systems. Conventional mel-frequency cepstral coefficients (MFCCs) are sensitive to noise of vibration signals, degrading classification accuracy. To solve this problem, this paper proposes spectral envelope cepstral coefficients (SECC) analysis, where a 4-step filter bank based on spectral envelopes of vibration signals is used: (1) a linear predictive coding (LPC) algorithm is used to specify spectral envelopes of all faulty vibration signals, (2) all envelopes are averaged to get general spectral shape, (3) a gradient descent method is used to find extremes of the average envelope and its frequencies, (4) a non-overlapped filter is used to have centers calculated from distances between valley frequencies of the envelope. This 4-step filter bank is then used in cepstral coefficients computation to extract feature vectors. Finally, a multi-layer support vector machine (MLSVM) with various sigma values uses these special parameters to identify faulty types of induction motors. Experimental results indicate that the proposed extraction method outperforms other feature extraction algorithms, yielding more than about 99.65% of classification accuracy.

• The Journal of the Acoustical Society of Korea
• /
• v.24 no.1
• /
• pp.29-37
• /
• 2005

This paper proposes an automatic genre classification system for Korean traditional music. The Proposed system accepts and classifies queried input music as one of the six musical genres such as Royal Shrine Music, Classcal Chamber Music, Folk Song, Folk Music, Buddhist Music, Shamanist Music based on music contents. In general, content-based music genre classification consists of two stages - music feature vector extraction and Pattern classification. For feature extraction. the system extracts 58 dimensional feature vectors including spectral centroid, spectral rolloff and spectral flux based on STFT and also the coefficient domain features such as LPC, MFCC, and then these features are further optimized using SFS method. For Pattern or genre classification, k-NN, Gaussian, GMM and SVM algorithms are considered. In addition, the proposed system adopts MFC method to settle down the uncertainty problem of the system performance due to the different query Patterns (or portions). From the experimental results. we verify the successful genre classification performance over $97{\%}$ for both the k-NN and SVM classifier, however SVM classifier provides almost three times faster classification performance than the k-NN.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.20 no.5
• /
• pp.421-427
• /
• 2009

This paper has proposed 1-2-1 coded cooperative communication that is a combination of STBC and ARQ. Coded cooperative communication is a protocol that integrates channel coding with cooperative communication. In this paper consider convolution encoder. ARQ method can increase the spectral efficiency than conventional cooperative communication because if the received signal from source node is satisfied by the destination preferentially, the destination transmits ACK message to both relay node and source node and then recovers the received signal. Where each relay 1, 2 forwards a punctured portion of receive data. When relay transmit to destination apply STBC the reliability to increase. Moreover this protocol can get better BER performance of receiver using simple comparator. We verified BER performance for the proposed protocol through Monte-Carlo simulation over Rayleigh fading plus AWGN.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.33 no.9C
• /
• pp.691-696
• /
• 2008

A new vocoder of Code Excited Linear Predictive (CELP) based on Adaptive Multi Rate (AMR) 7.4kbit/s mode is proposed in this paper. The proposed vocoder achieves a better compression rate in an environment of Speaker Dependent Coding System (SDSC) and is efficiently used for systems, such as OGM(Outgoing message) and TTS(Text To Speech), which needs only one person's speech. In order to enhance the compression rate of a coder, a new Line Spectral Pairs(LSP) code-book is employed by using Centroid Neural Network (CNN) algorithm. In comparison with original(traditional) AMR 7.4 Kbit/s coder, the new coder shows 27% higher compression rate while preserving synthesized speech quality in terms of Mean Opinion Score(MOS).

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.38 no.5
• /
• pp.576-582
• /
• 2001

The Multi-Band Excitation(MBE) speech coder uses a different approach for the representation of the excitation signal. It replaces the frame-based single voiced/unvoiced classification of a classical speech coder with a set of such decision over harmonic intervals in the frequency domain. This enables each speech segment to be a mixture of voiced and unvoiced, and improves the synthetic speech quality by reducing decision errors that might occur on the frame-based single voiced and unvoiced decision process when input speech is degraded with noise. The IMBE-LP, improved version of MBE with linear prediction, represents the spectral information of MBE model with linear prediction coefficients to obtain low bit rate of 2.4 kbps. In this Paper, we proposed a variable rate IMBE-LP vocoder that has lower bit rate than IMBE-LP without degrading the synthetic speech quality. To determine the LP order, it uses the spectral band information of the MBE model that has something to do with he input speech's characteristics. Experimental results are riven with our findings and discussions.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.32 no.3C
• /
• pp.234-240
• /
• 2007

In this paper, low complexity block constrained trellis coded quantization (BC-TCQ) structures are introduced, and a predictive BC TCQ encoding method is developed for quantization of line spectrum frequencies (LSF) parameters for narrowband speech coding applications. Trellis-coded quantization(TCQ) is a form of VQ that builds the VQ codebook from interleaved constituent scalar quantization codebooks. The performance is compared to the other VQ, demonstrating reduction in spectral distortion and significant reduction in encoding complexity. The predictive BC-TCQ is about 0.47107 dB superior to the IS-641 split-VQ, 26bits/frame, in spectral distortion sense. The BC-TCQ is 64.54%, 76.93%, 2.35% of the IS-641 split-VQ, respectively, in the complexity of the additions, multiplies, comparisons.