• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.43-51
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• 2005

In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a Performance of perceptual filter using noise energy control. Most of the algorithms which were proposed by the other researchers usually applied a filter using the noise energy acquired from a silent range. In this case. the improvement rate of tone quality decreases if the noise energy is changed by the magnitude or environment variation in a signal frame. But the Proposed method Provides the means to find a food estimated noise through energy control of the estimated noise which is obtained from a silent range. Also we can get the enhancement of tone qualify in low frequency band unlike other methods. To show the performance of the Proposed algorithm, various input signals which had a different signal-to-noise ratio (SNR) such as 5dB, l0dB, 15dB and 20dB were used to test the proposed algorithm. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR). noise-to-mask ration (NMR) and mean opinion score (MOS) test.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.98-101
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• 2000

On CELP type Vocoders G.723.1 6.3kbps/5.3kbps Dual Rate Speech Codec, which is developed for Internet Phone and videoconferencing, uses VAD(Voice Activity Detection)/CNG (Comfort Noise Generator) in order to reduce the bit rate in a silence period. In order to reduce the bit rate effectively in this paper, we first set the boundary condition of the energy threshold to prevent the consumption of unnecessary processing time, and use three decision rules to detect an active frame by energy, pitch gain and LSP distance. To evaluate the performance of the proposed algorithm we use silence-inserted speech data with 0, 5, 10, 20dB of SNR. As a result when SNR is over 5dB, the bit rate is reduced up to about 40% without speech degradation and the processing time is additionally decreased.

• Speech Sciences
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• v.15 no.2
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• pp.79-88
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• 2008

Recently demands for the speech technology-based products targeted for the mobile terminals such as cellular phones and PDA are rapidly increasing. And voting-based speaker identification algorithm is known to have a good performance in the mobile environment, since it works well with small amount of speaker training data. In this paper, we proposed a method to improve the performance of this voting based speaker identification system by using the observation confidence value which is derived from the function of SNR each frame. The proposed method is evaluated with ETRI cellular phone DB which is made for the speaker recognition task. The experimental results show that the proposed method has better performance of 2-3% identification rate than the conventional GMM method.

• Proceedings of the IEEK Conference
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• 2001.06a
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• pp.325-328
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• 2001

In this work, Cell search which is one of the important abilities of W-CDMA system in Reyleigh s fading channel is studied by using Cell Searcher of asynchronous IMT-2000 system(3GPP) and Cell - search simulation. For the methods of cell search to optimize codes, three stages are considered: 1) slot synchronization, 2) frame synchronization, and 3) scrambling code identification. It is found that key system parameters such as Primary Synchronization Channel (P-SCH), Secondary L Synchronization Channel(S-SCH), and Common Pilot Channel (CPCH) loading factor are optimized. It is noted that the smaller Optimal threshold value, the larger SNR of the received singnal. Therefore, It is important that the optimal threshold value is selected in the region of SNR

• Journal of Satellite, Information and Communications
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• v.10 no.2
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• pp.109-114
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• 2015

Frame synchronization is achieved by correlation between received symbols and a preamble pattern which is periodically appended at a frame header. In this paper, we deal with a frame detection method complaint with satellite-based DVB-S2 system. In DVB-S2, frame synchronization is performed under the low signal-to-noise ratio(SNR), a large frequency offset which can be up to 20% of a symbol transmission rate and unknown modulation schemes ranging from QPSK to 32-APSK. In this environment, we propose a method combining differential correlation based on SOF and PLSC with an accumulated correlation method for the detection of frame structures. In addition, detection performances about mean acquisition time(MAT) and detection error probability are evaluated via computer simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2027-2035
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• 2001

In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.7
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• pp.32-37
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• 2008

Satellite communication systems including the DVB-S2 (Digital Video Broadcasting - Satellite Version 2) system require operations under low signal-to-noise ratio (SNR) and large frequency offset values, and the initial frame synchronization process necessitates a robust correlation method. While a variety of conventional correlation structures exist for the initial synchronization, each method has different characteristics and performance in different channel environments. In this paper, we propose new correlation methods which exhibit enhanced performance in low SNR and large frequency offsets, and analyze their performance. The proposed methods use the magnitude sum and vector sum of extended differential correlation values, to maximize the correlation between the received signal and the synchronization sequence by using the spanned differential correlation result. The magnitude sum correlation method has better performance compared to conventional methods including the approximated ML (Maximum likelihood) method for SNR values below 4 dB with or without frequency offsets. The vector sum correlation method has improved performance over the magnitude sum method for channels with relatively small frequency offsets.

• Journal of the Institute of Convergence Signal Processing
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• v.12 no.3
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• pp.188-193
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• 2011

In this letter, we have derived another exact performance analysis for the OADF(opportunistic adaptive decode-and-forward) relay systems over Rayleigh fading channels. Based on error-events at relay nodes, the received instantaneous SNR(signal-to-noise ratio) is presented and its PDF(probability density function) is expressed as a more tractable form in which the number of summations and the length of each summation are specified. Then, exact average error rate, outage probability, and average channel capacity are obtained as general forms. Simulation results are finally presented to validate that the proposed analytical expressions can be a unified frame work covering all Rayleigh fading channel conditions. Furthermore, it is confirmed that OADF schemes can outperform the other schemes on the average error rate, outage probability, and average channel capacity.

• Journal of the Korea Society of Computer and Information
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• v.15 no.2
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• pp.109-117
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• 2010

Digital watermarking has drawn extensive attention for protecting digital contents from unauthorized copying. This paper proposes a new watermarking scheme in frequency domain for copyright protection of digital audio. In our proposed watermarking system, the original audio is segmented into non-overlapping frames. Watermarks are then embedded into the selected prominent peaks in the magnitude spectrum of each frame. Watermarks are extracted by performing the inverse operation of watermark embedding process. Simulation results indicate that the proposed scheme is robust against various kinds of attacks such as noise addition, cropping, resampling, re-quantization, MP3 compression, and low pass filtering. Our proposed watermarking system outperforms Cox's method in terms of imperceptibility, while keeping comparable robustness with the Cox's method. Our proposed system achieves SNR (signal-to-noise ratio) values ranging from 20 dB to 28 dB. This is in contrast to Cox's method which achieves SNR values ranging from only 14 dB to 23 dB.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.393-396
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• 2002

In this paper, we present a new method for generating a set of complementary sequences that can be simultaneously transmitted and yet compressed into a short pulse on reveive, and its application to ultrasound imaging. This new complementary sequences can be designed based on a filter bank theory. The new complementary sequences can be used to improve the SNR of ultrasound imaging without sacrificing the spatial resolution and frame rate, compared to conventional pulse-echo imaging. Computer simulations are performed to verify the proposed method.