• Journal of the Korean Institute of Telematics and Electronics
• /
• v.21 no.1
• /
• pp.71-78
• /
• 1984

In this paper a new method of improving the performance of ADPCM in noisy channel is proposed. The proposed method employs a robust quantizer, and transmits the information regarding the maximum step size periodically. Also, a scheme to correct most significant bit (MSB) errors is used in the receiver buffer. According to our computer simulation with real speech, the proposed ADPCM with error control yields an improvement of about 4 to 5 dB in noisy channel over the conventional ADPCM without error control.

• Journal of Communications and Networks
• /
• v.5 no.2
• /
• pp.157-166
• /
• 2003

This paper deals with the problem of channel identification for Single Input Multiple Output (SIMO) slow fading channels using clustering algorithms. Due to the intrinsic memory of the discrete-time model of the channel, over short observation periods, the received data vectors of the SIMO model are spread in clusters because of the AWGN noise. Each cluster is practically centered around the ideal channel output labels without noise and the noisy received vectors are distributed according to a multivariate Gaussian distribution. Starting from the Markov SIMO channel model, simultaneous maximum ikelihood estimation of the input vector and the channel coefficients reduce to one of obtaining the values of this pair that minimizes the sum of the Euclidean norms between the received and the estimated output vectors. Viterbi algorithm can be used for this purpose provided the trellis diagram of the Markov model can be labeled with the noiseless channel outputs. The problem of identification of the ideal channel outputs, which is the focus of this paper, is then equivalent to designing a Vector Quantizer (VQ) from a training set corresponding to the observed noisy channel outputs. The Linde-Buzo-Gray (LBG)-type clustering algorithms [1] could be used to obtain the noiseless channel output labels from the noisy received vectors. One problem with the use of such algorithms for blind time-varying channel identification is the codebook initialization. This paper looks at two critical issues with regards to the use of VQ for channel identification. The first has to deal with the applicability of this technique in general; we present theoretical results for the conditions under which the technique may be applicable. The second aims at overcoming the codebook initialization problem by proposing a novel approach which attempts to make the first phase of the channel estimation faster than the classical codebook initialization methods. Sample simulation results are provided confirming the effectiveness of the proposed initialization technique.

• Phonetics and Speech Sciences
• /
• v.4 no.4
• /
• pp.87-92
• /
• 2012

Speech enhancement aims to improve speech quality by removing background noise from noisy speech. Independent vector analysis is a type of frequency-domain independent component analysis method that is known to be free from the frequency bin permutation problem in the process of blind source separation from multi-channel inputs. This paper proposed a new method of microphone array based speech enhancement that combines independent vector analysis and beamforming techniques. Independent vector analysis is used to separate speech and noise components from multi-channel noisy speech, and delay-sum beamforming is used to determine the enhanced speech among the separated signals. To verify the effectiveness of the proposed method, experiments for computer simulated multi-channel noisy speech with various signal-to-noise ratios were carried out, and both PESQ and output signal-to-noise ratio were obtained as objective speech quality measures. Experimental results have shown that the proposed method is superior to the conventional microphone array based noise removal approach like GSC beamforming in the speech enhancement.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.6 no.8
• /
• pp.1161-1168
• /
• 2002

In this paper, the throughput performance of an ARQ scheme is analyzed with noisy feedback channel, in order to exactly analysis for performance of an ARQ scheme. An ARQ mechanism is presented by state diagram, and the throughput of an ARQ scheme is mathematically derived using generation function for a signal flow graph. The channel is modeled by the Nakagami-m fading channel which is distributed over far and wide, and the throughput performance of an ARQ scheme, which is applied in BPSK and BFSK systems, according to feedback gain and fading index m is analyzed through computer simulation. In the results, It is shown conformed that the throughput of an ARQ scheme decreased according to the increase of the noise for feedback channel, but it increased according to the increase of the feedback gain and fading index m. Also, it is shown that the throughput of an ARQ scheme using BPSK system is superior to BFSK system because of the difference of bit error probability between BPSK and BFSK systems.

• The Journal of the Acoustical Society of Korea
• /
• v.16 no.2
• /
• pp.41-48
• /
• 1997

This paper proposes the methods for improving the recognition rate of theARS, especially equipped with the speech recognition capability. Telephone speech, which is the input to the ARS, is usually affected by the announcements from the system, channel noise, and channel distortion, thus directly applying the recognition algorithm developed for clean speech to the noisy telephone speech will bring the significant performance degradation. To cope with this problem, this paper proposes three methods: 1)the accurate detection of the inputting instant of the speech in order to immediately turn off the announcements from the system at that instant, 2)the effective end-point detection of the noisy telephone speech on the basis of Teager energy, and 3)the SDCN-based compensation of the channel distortion. Experiments on speaker-independent, noisy telephone speech reveal that the combination of the above three proposed methods provides great improvements on the recognition rate over the conventional method, showing about 77% in contrast to only 23%.

• Journal of the Korean Institute of Telematics and Electronics B
• /
• v.31B no.4
• /
• pp.97-105
• /
• 1994

Transform image coding using DCT is proved to be efficient in the absence of channel error but its performance degrades rapidly over noisy channel. In this paper, in the case of appling bit selcetive error correction coding that protects some significant bits in a codeword, an efficient allocation method of imformation bits and additive redundancy bits used for quantization and error correction coding respectively under constant transmission bit rate is proposed, and its performance is analyzed. As a result, without increasing trasmission bit rate, PSNR can be improved up to 7~8 [dB] below bit error rate $10^2$ and the image without blocking effect caused by bit error resulted from channel noise can be recostructed.

• The Journal of the Acoustical Society of Korea
• /
• v.31 no.8
• /
• pp.487-506
• /
• 2012

This paper proposes a speech enhancement algorithm to improve the speech intelligibility by suppressing both reverberation and background noise. The algorithm adopts a non-causal single-channel minimum variance distortionless response (MVDR) filter to exploit an additional information that is included in the noisy-reverberant signals in subsequent frames. The noisy-reverberant signals are decomposed into the parts of the desired signal and the interference that is not correlated to the desired signal. Then, the filter equation is derived based on the MVDR criterion to minimize the residual interference without bringing speech distortion. The estimation of the correlation parameter, which plays an important role to determine the overall performance of the system, is mathematically derived based on the general statistical reverberation model. Furthermore, the practical implementation methods to estimate sub-parameters required to estimate the correlation parameter are developed. The efficiency of the proposed enhancement algorithm is verified by performance evaluation. From the results, the proposed algorithm achieves significant performance improvement in all studied conditions and shows the superiority especially for the severely noisy and strongly reverberant environment.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.42 no.3 s.303
• /
• pp.81-90
• /
• 2005

When we transmit signals, which are quantized by the vector quantizer (VQ), through noisy channels, the overall performance of the coding system is very dependent on the employed quantization scheme and the channel error effect. In order to design an optimal coding system, the source and channel coding scheme should be jointly optimized as in the channel-optimized VQ. As a suboptimal approach, we may consider the robust VQ (RVQ). In RVQ, we consider developing an index assignment function for mapping the output of quantizers to channel symbols so that the effect of the channel errors is minimized. Recently, a VQ, which can reduce the encoding complexity and is called the sample-adaptive product quantizer (SAPQ), has been proposed. SAPQ has very similar quantizer structure as to the product quantizer (PQ). However, the quantization performance can be better than PQ. Further, the encoding complexity and the memory requirement for the codebooks are lower than the regular full-search VQ case. In this paper, SAPQ is employed in order to design an RVQ to channel errors by reducing the vector dimension. Discussions on the codebook structure of SAPQ and experiments are introduced in an aspect of robustness to noisy channels.

• Phonetics and Speech Sciences
• /
• v.2 no.3
• /
• pp.141-148
• /
• 2010

In this paper, voice activity detection (VAD) for dual-channel noisy speech recognition is proposed in which spatial cues are employed. In the proposed method, a probability model for speech presence/absence is constructed using spatial cues obtained from dual-channel input signal, and a speech activity interval is detected through this probability model. In particular, spatial cues are composed of interaural time differences and interaural level differences of dual-channel speech signals, and the probability model for speech presence/absence is based on a Gaussian kernel density. In order to evaluate the performance of the proposed VAD method, speech recognition is performed for speech segments that only include speech intervals detected by the proposed VAD method. The performance of the proposed method is compared with those of several methods such as an SNR-based method, a direction of arrival (DOA) based method, and a phase vector based method. It is shown from the speech recognition experiments that the proposed method outperforms conventional methods by providing relative word error rates reductions of 11.68%, 41.92%, and 10.15% compared with SNR-based, DOA-based, and phase vector based method, respectively.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2015.05a
• /
• pp.160-163
• /
• 2015

현대인들은 필요한 물건들을 직접 구매하러 갈 시간이 부족하기 때문에 온라인 쇼핑몰의 이용 빈도가 늘어가고 있으며 이에 따라 온라인 쇼핑몰이 성행하고 있다. 하지만 온라인 쇼핑몰에서 물건을 구매하는 것은 물건을 눈으로 확인할 수 없다는 문제점이 있기 때문에 상품평은 구매를 결정하는데 많은 영향을 준다. 현재 온라인 쇼핑몰에서 고객이 상품평을 통해 상품에 대한 정보를 파악하기 어렵기 때문에 이를 해결하기 위한 연구들이 진행되고 있다. 이러한 연구들로 상품평의 의견을 분석하기 위한 연구로 오피니언 마이닝이 사용되고 있는 추세이다. 그러나 지금까지의 연구는 문법적인 오류, 신조어와 같이 국어사전에 등재되어 있지 않은 단어들을 감성분석기가 올바르게 판단하지 못하기 때문에 분석의 신뢰도가 떨어진다는 문제점이 있다. 그래서 형태소 분석을 실시하기 전에 신조어 사전을 추가하여 Noisy-channel model을 적용하여 더욱 정확한 감성분석이 가능하도록 하였다. 이러한 과정을 통해 가공된 정보를 바탕으로 상품평을 보다 정확하게 분석할 수 있는 시스템을 제안하고자 한다.