• Journal of Digital Contents Society
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• v.10 no.2
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• pp.233-238
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• 2009

The advances in the digital communication and network technology, Internet technology and the proliferation of smart appliances in home, have dramatically increased the need for a high speed/high quality home network. As consumer electronic devices and computing devices are increasing in the home network, it is obvious that the data traffic of home network increases as well. Various home network devices want to access Internet servers to get multimedia contents. Therefore, we introduce TLC(Telephone Line Carrier) system for networked digital consumer electronic appliances within a house using Ethernet or wire/wireless technology. In the future home network environment, the primary purposes of the smart home network based TLC are to create low-cost, easily deployable, high performance, and wide coverage throughout the home. In this paper, the channel capacity of telephone line communication system is evaluated and compared as a function of transmission power, number of OFDM carrier, channel loss, and noise loss for smart home network.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.41-48
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• 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 Biomedical Engineering Research
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• v.7 no.1
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• pp.21-34
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• 1986

In this paper, we described the ECG telemetry system via the Public Telephone Line. The system consist of a signal acquisition and measurement section, a signal processing section, and a signal transmission section. It used 8 bits microprocessor. The transmission section is composed of 3 ch. analog modulators and 1 ct. digital modem. Especially, using the digital modem, signal is transmitted with about 50n data reduction ratio by the TP (Turning Point) algorithm. The acoustic coupler or inductive coil for linking the public telephone line are used. The speed of the digital modem is 300 baud rate. The MCBS (Multi Channel Biotelemetry System) is tested and evaluated through the experiment.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.764-767
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• 2004

The performance of a speech recognition system is generally degraded in telephone environment because of distortions caused by background noise and various channel characteristics. In this paper, data-driven temporal filters are investigated to improve the performance of a specific recognition task such as telephone speech. Three different temporal filtering methods are presented with recognition results for Korean connected-digit telephone speech. Filter coefficients are derived from the cepstral domain feature vectors using the principal component analysis.

• MALSORI
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• no.53
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• pp.93-102
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• 2005

The performance of a speech recognition system is generally degraded in telephone environment because of distortions caused by background noise and various channel characteristics. In this paper, data-driven temporal filters are investigated to improve the performance of a specific recognition task such as telephone speech. Three different temporal filtering methods are presented with recognition results for Korean connected-digit telephone speech. Filter coefficients are derived from the cepstral domain feature vectors using the principal component analysis. According to experimental results, the proposed temporal filtering method has shown slightly better performance than the previous ones.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.3
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• pp.207-218
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• 1991

In this paper, the characteristics of MODEM with the low and medium transmission rate which are recommended by the international organization such as CCITT have been studied and the features of the telephone channel have been investigated. Then three kinds of classification algorithm for the modulated signals have been suggested and their performance have been researched through the computer simulation. In case that the channel is ideal, the modulated type such as FSK, PSK and QAM could be detected correctly. Moreover, when the modulated signals are received through the telpehone channel or noisy channel, it could be seen that the performance of the correlation algorithm is superior to that of other two algorithms because of the correlation being robust to the characteristics of telephone chanel and Gasussian noise, although the performances of the two classification algorithm using the peak and phase difference, signal constellation and so on decrease rapidly as the amplitude distortion, delay distortion and Gaussian noise increase.

• Speech Sciences
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• v.2
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• pp.57-66
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• 1997

This paper studies the effects of the method, CMS(Cepstral Mean Subtraction), (which compensates for some of the speech distortion. caused by telephone channels), on the performance of the text-dependent speaker identification system. This system is based on the VQ(Vector Quantization) and HMM(Hidden Markov Model) method and chooses the LPC-Cepstrum and Mel-Cepstrum as the feature vectors extracted from the speech data transmitted through telephone channels. Accordingly, we can compare the correct recognition rates of the speaker identification system between the use of LPC-Cepstrum and Mel-Cepstrum. Finally, from the experiment results table, it is found that the Mel-Cepstrum parameter is proven to be superior to the LPC-Cepstrum and that recognition performance improves by about 10% when compensating for telephone channel using the CMS.

• MALSORI
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• no.48
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• pp.93-105
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• 2003

Since telephone channel has bandlimited frequency characteristics, speech signal over the telephone channel shows degraded speech quality. In this paper, we propose an algorithm using neural network to reconstruct wideband speech from its narrowband version. Although single neural network is a good tool for direct mapping, it has difficulty in training for vast and complicated data. To alleviate this problem, we modularize the neural networks based on appropriate clustering of the acoustic space. We also introduce fuzzy computing to compensate for probable misclassification at the cluster boundaries. According to our simulation, the proposed algorithm showed improved performance over the single neural network and conventional codebook mapping method in both objective and subjective evaluations.

• MALSORI
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• no.45
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• pp.93-105
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• 2003

In this paper, we present a single framework based on GMM for speaker recognition. The proposed framework can simultaneously minimize environmental variations on mismatched conditions and adapt the bias free and speaker-dependent characteristics of claimant utterances to the background GMM to create a speaker model. We compare the closed-set speaker identification for conventional method and the proposed method both on TIMIT and NTIMIT. In the several sets of experiments we show the improved recognition rates on a simulated channel and a telephone channel condition by 7.2% and 27.4% respectively.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1567-1575
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• 1996

This paper presents a new power control method for compensating the short term fading of the reverse link channel in the CDMA mobile telephone system. The fixed step closed loop power control which is now adopted in IS-95, is very simple in structure. However, the step size in the closed loop power control is too big for the channel with a small variation or too big for the channel with a small variation or too small for the channel with a large variation. The method presented in this paper has a simple structure and shows a new model employing the combination of the fixed step size method and variable step size method which results in compensatingthe disadvantages mentioned above. This paper also evaluates the performance inthe fundamental channel model.