Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364793
S. Chai, David Wu, D. Tan, Henry Wu, D. Tan, Hong Ren Wu
Perceptually lossless medical image coding is an alternative for compressing medical images. It provides a solution to the challenge of delivering clinically critical information in the shortest time possible. In this paper, an investigation of the robustness of the perceptually lossless image coder is carried out by applying bilinear, biquadratic and bicubic standard and centered B-spline interpolation filters. In order to evaluate the visual performance, a subjective assessment was conducted consisting of 30 medical images and 6 image processing experts. Here, the perceptually lossless medical image coder was compared to the state-of-the-art JPEG-LS compliant LOCO and NLOCO image coders. Current results have shown that overall, there were no perceivable differences of statistical significance when the medical images were enlarged by a factor of 2
{"title":"Robustness of a Perceptually Lossless Medical Image Coder: Interpolation","authors":"S. Chai, David Wu, D. Tan, Henry Wu, D. Tan, Hong Ren Wu","doi":"10.1109/ISPACS.2006.364793","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364793","url":null,"abstract":"Perceptually lossless medical image coding is an alternative for compressing medical images. It provides a solution to the challenge of delivering clinically critical information in the shortest time possible. In this paper, an investigation of the robustness of the perceptually lossless image coder is carried out by applying bilinear, biquadratic and bicubic standard and centered B-spline interpolation filters. In order to evaluate the visual performance, a subjective assessment was conducted consisting of 30 medical images and 6 image processing experts. Here, the perceptually lossless medical image coder was compared to the state-of-the-art JPEG-LS compliant LOCO and NLOCO image coders. Current results have shown that overall, there were no perceivable differences of statistical significance when the medical images were enlarged by a factor of 2","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121975404","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364914
C. Chu, N. Wong
This paper presents a highly efficient approach for performing positive real balanced truncation (PRBT) of single-input single-output (SISO) systems. The solution of two dual algebraic Riccati equations (AREs), whose high computational cost baffles conventional PRBT, is replaced with the solution of a cross-Riccati equation (CRE). The cross-Riccatian nature of the solution then allows simple construction of the projection matrices in PRBT. Application examples confirm the effectiveness of the proposed PRBT method over conventional schemes
{"title":"Efficient Positive Real Balanced Truncation of SISO Systems Via Cross-Riccati Equations","authors":"C. Chu, N. Wong","doi":"10.1109/ISPACS.2006.364914","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364914","url":null,"abstract":"This paper presents a highly efficient approach for performing positive real balanced truncation (PRBT) of single-input single-output (SISO) systems. The solution of two dual algebraic Riccati equations (AREs), whose high computational cost baffles conventional PRBT, is replaced with the solution of a cross-Riccati equation (CRE). The cross-Riccatian nature of the solution then allows simple construction of the projection matrices in PRBT. Application examples confirm the effectiveness of the proposed PRBT method over conventional schemes","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"24 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125006734","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364780
Chi-Min Li, Jia-Chyi Wu, I. Tang
Wideband code division multiple access (W-CDMA) adopts the smart antenna techniques to increase the signal-to-interference-noise ratio (SINR) and system capacity. In this paper, we statistically derive an analytic result to prove the SINR performance for the two commonly used CC and DOA beamforming methods. Results show the both methods will have the same mean SINR performance and CC method will be more robust than the DOA method.
{"title":"An Analytic Analysis of W-CDMA Smart Antennas Beamforming Using Complex Conjugate and DOA Methods","authors":"Chi-Min Li, Jia-Chyi Wu, I. Tang","doi":"10.1109/ISPACS.2006.364780","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364780","url":null,"abstract":"Wideband code division multiple access (W-CDMA) adopts the smart antenna techniques to increase the signal-to-interference-noise ratio (SINR) and system capacity. In this paper, we statistically derive an analytic result to prove the SINR performance for the two commonly used CC and DOA beamforming methods. Results show the both methods will have the same mean SINR performance and CC method will be more robust than the DOA method.","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"23 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122124850","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364831
S. Tsuge, Minoru Fukumi, M. Shishibori, Fuji Ren, K. Kita, S. Kuroiwa
Even if a speaker uses a speaker-dependent speech recognition system, speech recognition performance varies. For this reason, speech quality is varied by some factors, including emotion, background noise, and so on, even though the speaker and utterance remain constant. However, the relationships between intra-speaker's speech variability and speech recognition performance are not clear. Hence, we focus on the intra-speaker's speech variability which affects the speech recognition performances. To investigate these relationships, we have been collecting speech data since November 2002. Using a part of the speech corpus, we conducted speech recognition experiments. In this paper, we analyze the relationships between intra-speaker's speech variability and the phoneme accuracy by using the correlation analysis. For factors of the correlation analysis, we use a number of errors, a speaking rate, a likelihood. Analysis results show a strong correlation between the number of the substitution errors and the phoneme accuracy although the correlations of the number of the deletion and the insertion errors are low. Therefore, it is considered that there are overlaps between phonemes since the feature parameters vary at each speaking rate. For improving the phoneme accuracy, it is needed that we study a method which discriminates phonemes. On the other hand, although the correlation between the phoneme accuracy and the speaking rate seems to be low, a strong correlation between the speaking rate and the number of deletion errors and insertion errors are found. Since the number of the insertion errors and the number of the deletion errors were in the counterbalance relation, the correlation between the speaking rate and the phoneme accuracy was low. However, we consider that it is needed to normalize the speaking rate because the speaking rate influences on the number of the deletion and the insertion errors
{"title":"Study of Relationships between Intra-speaker's Speech Variability and Speech Recognition Performance","authors":"S. Tsuge, Minoru Fukumi, M. Shishibori, Fuji Ren, K. Kita, S. Kuroiwa","doi":"10.1109/ISPACS.2006.364831","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364831","url":null,"abstract":"Even if a speaker uses a speaker-dependent speech recognition system, speech recognition performance varies. For this reason, speech quality is varied by some factors, including emotion, background noise, and so on, even though the speaker and utterance remain constant. However, the relationships between intra-speaker's speech variability and speech recognition performance are not clear. Hence, we focus on the intra-speaker's speech variability which affects the speech recognition performances. To investigate these relationships, we have been collecting speech data since November 2002. Using a part of the speech corpus, we conducted speech recognition experiments. In this paper, we analyze the relationships between intra-speaker's speech variability and the phoneme accuracy by using the correlation analysis. For factors of the correlation analysis, we use a number of errors, a speaking rate, a likelihood. Analysis results show a strong correlation between the number of the substitution errors and the phoneme accuracy although the correlations of the number of the deletion and the insertion errors are low. Therefore, it is considered that there are overlaps between phonemes since the feature parameters vary at each speaking rate. For improving the phoneme accuracy, it is needed that we study a method which discriminates phonemes. On the other hand, although the correlation between the phoneme accuracy and the speaking rate seems to be low, a strong correlation between the speaking rate and the number of deletion errors and insertion errors are found. Since the number of the insertion errors and the number of the deletion errors were in the counterbalance relation, the correlation between the speaking rate and the phoneme accuracy was low. However, we consider that it is needed to normalize the speaking rate because the speaking rate influences on the number of the deletion and the insertion errors","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124679347","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364788
Peng Shao, Y. Tanada
A real-valued self-orthogonal finite-length sequence has a zero-sidelobe autocorrelation function except at both shift ends. Two-valued and three-valued integrand codes for the sequence are effective for a practical application. This paper presents the improved two-valued and three-valued integrand codes. The allocation of code values in every time chip is optimized so that the Fourier spectrum of the equivalent multivalued pulse-width-modulation (PWM) code waveform at every chip may resemble each other. The codes are smoothed through Gaussian low pass filter. The sidelobes of the autocorrelation outputs are improved to be about 10-dB lower than those for the not-optimized codes.
{"title":"Improved Two-Valued and Three-Valued Integrand Codes for Real-Valued Self-Orthogonal Finite-Length Sequence","authors":"Peng Shao, Y. Tanada","doi":"10.1109/ISPACS.2006.364788","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364788","url":null,"abstract":"A real-valued self-orthogonal finite-length sequence has a zero-sidelobe autocorrelation function except at both shift ends. Two-valued and three-valued integrand codes for the sequence are effective for a practical application. This paper presents the improved two-valued and three-valued integrand codes. The allocation of code values in every time chip is optimized so that the Fourier spectrum of the equivalent multivalued pulse-width-modulation (PWM) code waveform at every chip may resemble each other. The codes are smoothed through Gaussian low pass filter. The sidelobes of the autocorrelation outputs are improved to be about 10-dB lower than those for the not-optimized codes.","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"40 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129573467","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364801
N. Punthong, A. Surarerks
An encoding technique is an interesting problem in information theory (i.e., how to reduce the size of the textual material when information is continually sent from one point to another.) The important factors to improve efficiency of the encoding process are probability distribution and dependency of the input data. Some encoding algorithms focused on these factors are Huffman algorithm, Lempel-Ziv algorithm and enhance versions of them. In some cases, dependency of the input data is not significant, but the probability distribution remains considerable. In this paper, we propose a novel approach for an on-line encoding algorithm using a generating function that the encoding process performs in an on-line manner. Our concept is that the generating function must be constructed up to the probability distribution. Some experimental results show that our technique can apply to the normal distribution of input data
{"title":"A Novel Approach for On-line Encoding Algorithm Using A Generating Function","authors":"N. Punthong, A. Surarerks","doi":"10.1109/ISPACS.2006.364801","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364801","url":null,"abstract":"An encoding technique is an interesting problem in information theory (i.e., how to reduce the size of the textual material when information is continually sent from one point to another.) The important factors to improve efficiency of the encoding process are probability distribution and dependency of the input data. Some encoding algorithms focused on these factors are Huffman algorithm, Lempel-Ziv algorithm and enhance versions of them. In some cases, dependency of the input data is not significant, but the probability distribution remains considerable. In this paper, we propose a novel approach for an on-line encoding algorithm using a generating function that the encoding process performs in an on-line manner. Our concept is that the generating function must be constructed up to the probability distribution. Some experimental results show that our technique can apply to the normal distribution of input data","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130954763","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364783
Koksheik Wong, K. Tanaka, Xiaojun Qi
This paper presents a graphical comparative study of steganographic methods in the DCT domain. This graphical representation allows at-one-glance comparison among steganographic methods, showing the relative performance of each method with respect to the ideal methods for the metrics considered. Six representative DCT-based steganographic methods are selected and fairly compared in terms of six significant evaluation criteria. From the comparison results, we found some trends in recent evolution of steganographic methods and possible future development directions.
{"title":"Graphical Comparative Study on DCT-based Steganographic Methods","authors":"Koksheik Wong, K. Tanaka, Xiaojun Qi","doi":"10.1109/ISPACS.2006.364783","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364783","url":null,"abstract":"This paper presents a graphical comparative study of steganographic methods in the DCT domain. This graphical representation allows at-one-glance comparison among steganographic methods, showing the relative performance of each method with respect to the ideal methods for the metrics considered. Six representative DCT-based steganographic methods are selected and fairly compared in terms of six significant evaluation criteria. From the comparison results, we found some trends in recent evolution of steganographic methods and possible future development directions.","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132322673","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364895
N. Tanabe, T. Furukawa, H. Matsue, S. Tsujii
We propose a noise suppression algorithm based on Kaiman filter. The algorithm achieves a noise suppression with high speech quality under the condition of the AWGN (additive white Gaussian noise), from the canonical state space models with a state equation composed of the clean speech signal and an observation equation composed of the clean speech signal and AWGN. The special feature of the proposed algorithm is realization of high speech quality noise suppression utilizing only the Kalman filter, while the conventional algorithm utilizes the linear prediction algorithm and the Kalman filter. The simulation results show that the proposed method improved the noise suppression capability by about 5 dB than that of the conventional method
提出了一种基于Kaiman滤波的噪声抑制算法。该算法从由纯净语音信号组成的状态方程和由纯净语音信号和加性高斯白噪声组成的观测方程的正则状态空间模型出发，实现了在加性高斯白噪声条件下的高质量语音抑制。该算法的特点是仅利用卡尔曼滤波器实现高语音质量的噪声抑制，而传统算法则利用线性预测算法和卡尔曼滤波器。仿真结果表明，该方法的噪声抑制能力比传统方法提高了约5 dB
{"title":"Noise Suppression with High Speech Quality Based on Kalman Filter","authors":"N. Tanabe, T. Furukawa, H. Matsue, S. Tsujii","doi":"10.1109/ISPACS.2006.364895","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364895","url":null,"abstract":"We propose a noise suppression algorithm based on Kaiman filter. The algorithm achieves a noise suppression with high speech quality under the condition of the AWGN (additive white Gaussian noise), from the canonical state space models with a state equation composed of the clean speech signal and an observation equation composed of the clean speech signal and AWGN. The special feature of the proposed algorithm is realization of high speech quality noise suppression utilizing only the Kalman filter, while the conventional algorithm utilizes the linear prediction algorithm and the Kalman filter. The simulation results show that the proposed method improved the noise suppression capability by about 5 dB than that of the conventional method","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"67 5","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132949050","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
This paper presents a common gate low-power low noise amplifier (LNA) for DS-UWB low-band and MB-OFDM group-A ultra-wideband radio system. The LNA is designed and implemented in 0.18 mum RF CMOS process. Simulation results show that power gain of 15.4 dB, input and output matching lower than -10.6 dB and -13.3 dB, and a minimum NF of 3.8 dB can be achieved, while the power consumption is only 3.24 mW through 1.8 V power supply
{"title":"A Low-Power and Low-Noise Amplifier for 3-5GHz UWB Applications","authors":"Chun-Chieh Chen, Sheng-Hsiang Yen, Zhe-Yang Huang, Meng-Ping Chen, Yeh-Tai Hung","doi":"10.1109/ISPACS.2006.364864","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364864","url":null,"abstract":"This paper presents a common gate low-power low noise amplifier (LNA) for DS-UWB low-band and MB-OFDM group-A ultra-wideband radio system. The LNA is designed and implemented in 0.18 mum RF CMOS process. Simulation results show that power gain of 15.4 dB, input and output matching lower than -10.6 dB and -13.3 dB, and a minimum NF of 3.8 dB can be achieved, while the power consumption is only 3.24 mW through 1.8 V power supply","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"33 4","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114119357","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364705
Jeng-Kuang Hwang, Chen-Yu Chen, R. Chung
A novel linearized Gaussian minimum shift keying (LGMSK) system is proposed for block data transmission. In each block, a known unique-word (UW) is inserted as guard interval, and a phase state return bit (PSRB) is also added to maintain LGMSK phase trellis consistency. It is shown that the transmit signal has near constant-envelope property. At the receiver, a frequency-domain decision-feedback-equalizer (FD-DFE) is designed to combat the joint effect of partial response signaling and the channel dispersion. With only a few feedback taps, the receiver can significantly outperform the FD linear equalizer. Compared with the Viterbi receiver, it has only 0.4-dB degradation at BER=10-6 over AWGN channel. Hence, the proposed system has the great merits of transmit power saving, high performance, and low receiver complexity, especially for high-rate application through long dispersion channel
{"title":"Unique-Word-Aided Linearized GMSK System with Frequency-Domain Decision Feedback Equalizer","authors":"Jeng-Kuang Hwang, Chen-Yu Chen, R. Chung","doi":"10.1109/ISPACS.2006.364705","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364705","url":null,"abstract":"A novel linearized Gaussian minimum shift keying (LGMSK) system is proposed for block data transmission. In each block, a known unique-word (UW) is inserted as guard interval, and a phase state return bit (PSRB) is also added to maintain LGMSK phase trellis consistency. It is shown that the transmit signal has near constant-envelope property. At the receiver, a frequency-domain decision-feedback-equalizer (FD-DFE) is designed to combat the joint effect of partial response signaling and the channel dispersion. With only a few feedback taps, the receiver can significantly outperform the FD linear equalizer. Compared with the Viterbi receiver, it has only 0.4-dB degradation at BER=10-6 over AWGN channel. Hence, the proposed system has the great merits of transmit power saving, high performance, and low receiver complexity, especially for high-rate application through long dispersion channel","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114399060","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
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