Pub Date : 2016-10-01DOI: 10.1109/ISPACS.2016.7824752
Piyawit Tantisarkhornkhet, Warodom Werapun, B. Paillassa
The Prince of Songkla University (PSU) is the top five universities of Thailand that still uses traditional networking. Presently, there are Internet applications that are able to provide dynamic, manageable, and adaptable features such as Software-defined Networking (SDN). SDN is a recent concept of programmable networks that divides the control plane and data plane of all network devices. It can be programmed via an open interface which is interesting to implement SDN because of their various benefits such as centralized network provisioning, lower operating costs etc. In this paper, we propose comparison between traditional PSU network which is defined Static routing (Static routing non-SDN) and SDN which are using programmable algorithm such as Bellman-Ford SDN (BFSDN) unicast, Dijkstra SDN (DSDN) on both unicast and multicast in order to determine worthiness of migration from traditional network to SDN. In part of topology emulator, we have replicated topology by Mininet. Its performance is examined in terms of throughput, latency, jitter, and packet loss.
宋卡王子大学(PSU)是泰国仍在使用传统网络的前五所大学。目前，有一些Internet应用程序能够提供动态的、可管理的和可适应的特性，例如软件定义网络(SDN)。SDN是一种新的可编程网络概念，它将所有网络设备的控制平面和数据平面分开。它可以通过一个开放的接口进行编程，这对于实现SDN来说很有趣，因为它有各种各样的好处，比如集中的网络供应，更低的运营成本等。本文对传统的PSU网络进行了静态路由(静态路由非SDN)和使用可编程算法的SDN(如Bellman-Ford SDN (BFSDN)单播、Dijkstra SDN (DSDN)单播和多播)的比较，以确定从传统网络迁移到SDN的价值。在拓扑模拟器的一部分，我们用Mininet复制了拓扑。它的性能是根据吞吐量、延迟、抖动和数据包丢失来检查的。
{"title":"SDN experimental on the PSU network","authors":"Piyawit Tantisarkhornkhet, Warodom Werapun, B. Paillassa","doi":"10.1109/ISPACS.2016.7824752","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824752","url":null,"abstract":"The Prince of Songkla University (PSU) is the top five universities of Thailand that still uses traditional networking. Presently, there are Internet applications that are able to provide dynamic, manageable, and adaptable features such as Software-defined Networking (SDN). SDN is a recent concept of programmable networks that divides the control plane and data plane of all network devices. It can be programmed via an open interface which is interesting to implement SDN because of their various benefits such as centralized network provisioning, lower operating costs etc. In this paper, we propose comparison between traditional PSU network which is defined Static routing (Static routing non-SDN) and SDN which are using programmable algorithm such as Bellman-Ford SDN (BFSDN) unicast, Dijkstra SDN (DSDN) on both unicast and multicast in order to determine worthiness of migration from traditional network to SDN. In part of topology emulator, we have replicated topology by Mininet. Its performance is examined in terms of throughput, latency, jitter, and packet loss.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":" 9","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"113949808","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 : 2016-10-01DOI: 10.1109/ISPACS.2016.7824748
In-Gul Jang, Gweon-Do Jo
In this paper, we propose latency efficient Inverse Fast Fourier Transform (IFFT) design method reducing the latency of IFFT output through the reordering of IFFT input data from the resource element mapper to IFFT input signal. The IFFT core consumes a significant percentage for high speed communication systems such as Long Term Evolution (LTE). So, IFFT processor in the physical layer implementations of baseband modem which is important component since IFFT processors require large amount of area and processing power. Also, IFFT has quite long latency from IFFT input data to output data. Therefore latency efficient IFFT is needed for providing various applications such as real time service without latency. Proposed IFFT architecture reduces IFFT output data delay through the reduction of IFFT memory size and butterfly operation (e.g. addition / subtraction). Third Generation Partnership Project - Long Term Evolution (3GPP - LTE) systems use 2048-point FFT processor in the 20MHz bandwidth. Thus, input signal of the IFFT processor corresponding to guard band are assigned as null (‘0’). Based on the fact that there are many null as an input signals of IFFT, a hardware and latency efficient IFFT design method for low latency communication systems like 5G LTE is proposed. To verify the performance of the proposed algorithm, 2048 point FFT with radix-2 based SDF structure is used.
{"title":"Study on the latency efficient IFFT design method for low latency communication systems","authors":"In-Gul Jang, Gweon-Do Jo","doi":"10.1109/ISPACS.2016.7824748","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824748","url":null,"abstract":"In this paper, we propose latency efficient Inverse Fast Fourier Transform (IFFT) design method reducing the latency of IFFT output through the reordering of IFFT input data from the resource element mapper to IFFT input signal. The IFFT core consumes a significant percentage for high speed communication systems such as Long Term Evolution (LTE). So, IFFT processor in the physical layer implementations of baseband modem which is important component since IFFT processors require large amount of area and processing power. Also, IFFT has quite long latency from IFFT input data to output data. Therefore latency efficient IFFT is needed for providing various applications such as real time service without latency. Proposed IFFT architecture reduces IFFT output data delay through the reduction of IFFT memory size and butterfly operation (e.g. addition / subtraction). Third Generation Partnership Project - Long Term Evolution (3GPP - LTE) systems use 2048-point FFT processor in the 20MHz bandwidth. Thus, input signal of the IFFT processor corresponding to guard band are assigned as null (‘0’). Based on the fact that there are many null as an input signals of IFFT, a hardware and latency efficient IFFT design method for low latency communication systems like 5G LTE is proposed. To verify the performance of the proposed algorithm, 2048 point FFT with radix-2 based SDF structure is used.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":"97 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114518549","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 : 2016-10-01DOI: 10.1109/ISPACS.2016.7824724
Takahide Sato, Ong Zi Hao, S. Kasai
A cascaded differential 20-path and 3-path filter which realizes a more than 20 dB stopband rejection and harmonic rejection ability is proposed. Compared to a differential N-path filters with a passive low-pass filter connected between the signal source and the filter itself, the proposed cascaded N-path filter provides better flexibility in tuning and narrower full width half maximum (FWHM) value for higher frequency selectivity. The first stage 20-path filter which acts as a tunable low frequency band-pass filter operating at one third of switching frequency of second stage N-path filter, attenuating the undesirable high frequency signals. The second stage 3-path filter then filters off the remaining standards providing sufficient stopband rejection for the whole system. The different operating frequencies for both stages provides a harmonic rejection ability as the high frequency components causing the harmonic folding of second stage frequency are first removed before passing through the second stage N-path filter for a desirable standard.
{"title":"A distinct operation frequency cascaded N-path filter with improved harmonic rejection ability","authors":"Takahide Sato, Ong Zi Hao, S. Kasai","doi":"10.1109/ISPACS.2016.7824724","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824724","url":null,"abstract":"A cascaded differential 20-path and 3-path filter which realizes a more than 20 dB stopband rejection and harmonic rejection ability is proposed. Compared to a differential N-path filters with a passive low-pass filter connected between the signal source and the filter itself, the proposed cascaded N-path filter provides better flexibility in tuning and narrower full width half maximum (FWHM) value for higher frequency selectivity. The first stage 20-path filter which acts as a tunable low frequency band-pass filter operating at one third of switching frequency of second stage N-path filter, attenuating the undesirable high frequency signals. The second stage 3-path filter then filters off the remaining standards providing sufficient stopband rejection for the whole system. The different operating frequencies for both stages provides a harmonic rejection ability as the high frequency components causing the harmonic folding of second stage frequency are first removed before passing through the second stage N-path filter for a desirable standard.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116063335","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 : 2016-10-01DOI: 10.1109/ISPACS.2016.7824685
Felix Labelle, R. Lefebvre, P. Gournay
The accuracy of the reproduction of emotions by speech coders has only recently been identified as a relevant issue. Several published studies have shown that speech compression reduces the accuracy of emotions classification. These studies, however, were all conducted using objective evaluation methods that involve an automatic classifier. The only definitive way to prove or disprove that the emotional content of a speech signal is degraded by compression operations is by testing it with human subjects. This paper proposes a subjective evaluation method and applies it to emotional speech coded by the AMR-WB speech coder at 6.6 and 12.65 kbps. The results confirm that there is a significant degradation in the perception of emotions by human listeners at both bit rates. The proposed evaluation method, and the insight provided by the results, could be useful in developing new speech coders that better preserve the emotional content of speech signals.
{"title":"A subjective evaluation of the effects of speech coding on the perception of emotions","authors":"Felix Labelle, R. Lefebvre, P. Gournay","doi":"10.1109/ISPACS.2016.7824685","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824685","url":null,"abstract":"The accuracy of the reproduction of emotions by speech coders has only recently been identified as a relevant issue. Several published studies have shown that speech compression reduces the accuracy of emotions classification. These studies, however, were all conducted using objective evaluation methods that involve an automatic classifier. The only definitive way to prove or disprove that the emotional content of a speech signal is degraded by compression operations is by testing it with human subjects. This paper proposes a subjective evaluation method and applies it to emotional speech coded by the AMR-WB speech coder at 6.6 and 12.65 kbps. The results confirm that there is a significant degradation in the perception of emotions by human listeners at both bit rates. The proposed evaluation method, and the insight provided by the results, could be useful in developing new speech coders that better preserve the emotional content of speech signals.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":"59 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114705727","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 : 2016-10-01DOI: 10.1109/ISPACS.2016.7824753
P. Phatsornsiri, M. Kumngern, F. Khateb
This paper presents a new first-order allpass filter using bulk-driven transconductor which is suitable for biological signal processing applications. The proposed filter employs one transconductor, one resistor and two capacitors. The bulk-driven MOS transistor technique is used to provide 0.5 V supply voltage operation. The workability of the proposed topology is expressed through PSPICE simulators using TSMC 0.18 µm n-well CMOS process. Simulation results show that the circuit consumes the power of 11.7 µW and has total harmonic distortion (THD) of 1% for input signal of 30 mVP-P.
提出了一种适用于生物信号处理的一阶全通滤波器。所提出的滤波器采用一个晶体管、一个电阻器和两个电容器。采用体积驱动MOS晶体管技术提供0.5 V的供电电压。采用台积电0.18 μ m n阱CMOS工艺，通过PSPICE模拟器表达了所提出拓扑的可操作性。仿真结果表明，当输入信号为30 mVP-P时，电路功耗为11.7µW，总谐波失真(THD)为1%。
{"title":"0.5-V fully differential allpass section","authors":"P. Phatsornsiri, M. Kumngern, F. Khateb","doi":"10.1109/ISPACS.2016.7824753","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824753","url":null,"abstract":"This paper presents a new first-order allpass filter using bulk-driven transconductor which is suitable for biological signal processing applications. The proposed filter employs one transconductor, one resistor and two capacitors. The bulk-driven MOS transistor technique is used to provide 0.5 V supply voltage operation. The workability of the proposed topology is expressed through PSPICE simulators using TSMC 0.18 µm n-well CMOS process. Simulation results show that the circuit consumes the power of 11.7 µW and has total harmonic distortion (THD) of 1% for input signal of 30 mVP-P.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":"22 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124051500","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 : 2016-10-01DOI: 10.1109/ISPACS.2016.7824686
K. Imamura, Naoki Kimura, Fumiaki Satou, S. Sanada, Y. Matsuda
The non-local means method is a high-performance noise reduction method that utilizes the structural similarity of an image. The non-local means method generally assumes the noise is Gaussian, and the noise strength is distributed evenly over an image. In the normal non-local means, the weighting function for the noise reduction strength is controlled by a single fixed parameter. However, the non-local means method is not suitable for application to X-ray images, due to the existence of Poisson noise, in its current form. In this paper, we propose an image denoising method using non-local means for an image with Poisson noise. The weighting function in the proposed method adjusts the weight parameter based on the estimated noise strength from the pixels in a local region. As a result, the proposed method provides good noise reduction performance for Poisson noise without recourse to a variance stabilizing transformation. We demonstrate that the noise reduction of the proposed method is an improvement of 0.1–0.9 dB compared to the standard non-local means.
{"title":"Image denoising using non-local means for Poisson noise","authors":"K. Imamura, Naoki Kimura, Fumiaki Satou, S. Sanada, Y. Matsuda","doi":"10.1109/ISPACS.2016.7824686","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824686","url":null,"abstract":"The non-local means method is a high-performance noise reduction method that utilizes the structural similarity of an image. The non-local means method generally assumes the noise is Gaussian, and the noise strength is distributed evenly over an image. In the normal non-local means, the weighting function for the noise reduction strength is controlled by a single fixed parameter. However, the non-local means method is not suitable for application to X-ray images, due to the existence of Poisson noise, in its current form. In this paper, we propose an image denoising method using non-local means for an image with Poisson noise. The weighting function in the proposed method adjusts the weight parameter based on the estimated noise strength from the pixels in a local region. As a result, the proposed method provides good noise reduction performance for Poisson noise without recourse to a variance stabilizing transformation. We demonstrate that the noise reduction of the proposed method is an improvement of 0.1–0.9 dB compared to the standard non-local means.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":"141-142 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"117172157","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 : 2016-10-01DOI: 10.1109/ISPACS.2016.7824749
Artit Visavakitcharoen, Yuma Kinoshita, H. Kobayashi, H. Kiya
This paper proposes a quality improvement method of tone mapped images including compression distortion for the JPEG XT standard, which is a new compression standard for high dynamic range (HDR) images. HDR images are generally required to be mapped to low dynamic range (LDR) ones due to the limitation of display devices. Furthermore, the HDR ones include some compression distortion to be efficiently stored, in most cases. However, conventional tone mapping operations have not considered the effect of the distortion. We apply an iterative gradient ascend algorithm for improving the structural fidelity, which is based on the improved tone mapped image quality index (TMQI-II), under the use of the JPEG XT standard. Decompressed algorithm and the results are compared with those of original HDR ones. The experiment confirms that the images with better quality than images without optimization are provided by the proposed method and the quality is close to that of ones generated from the original HDR image without compression distortion.
{"title":"Quality improvement of tone mapped images by TMQI-II based optimization for the JPEG XT standard","authors":"Artit Visavakitcharoen, Yuma Kinoshita, H. Kobayashi, H. Kiya","doi":"10.1109/ISPACS.2016.7824749","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824749","url":null,"abstract":"This paper proposes a quality improvement method of tone mapped images including compression distortion for the JPEG XT standard, which is a new compression standard for high dynamic range (HDR) images. HDR images are generally required to be mapped to low dynamic range (LDR) ones due to the limitation of display devices. Furthermore, the HDR ones include some compression distortion to be efficiently stored, in most cases. However, conventional tone mapping operations have not considered the effect of the distortion. We apply an iterative gradient ascend algorithm for improving the structural fidelity, which is based on the improved tone mapped image quality index (TMQI-II), under the use of the JPEG XT standard. Decompressed algorithm and the results are compared with those of original HDR ones. The experiment confirms that the images with better quality than images without optimization are provided by the proposed method and the quality is close to that of ones generated from the original HDR image without compression distortion.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":"94 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115203112","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 : 2016-10-01DOI: 10.1109/ISPACS.2016.7824699
Imdad MaungMaung, Koksheik Wong, Kiyoshi Tanaka
In this work, two reversible data hiding methods are proposed by exploiting the audio-video synchronization process in the MP4 container. Specifically, audio-video synchronization information is stored as time-to-sample information in the stts box of the MP4 container. In Method 1, the number of sample counts in the stts box is decomposed into multiple integers, where each integer represents the decimal equivalent of the payload (represented in binary) to be hidden. Method 1 completely preserves the audio-video synchronization before and after hiding data into the MP4 container. To suppress bitstream size increment and improve the number of bits that can be hidden, Method 2 is proposed by manipulating both timescale and duration of sample δ to hide data. Basic performance of the proposed methods are verified through experiments on various H.264/AVC and AAC compressed short movie clips downloaded from YouTube. In the worst case scenario, a negligible bitstream size increment of < 0.0447% and absolute synchronization error of < 13 milliseconds were observed when payload of length 1024 bits is embedded into the movie clips.
{"title":"Reversible data hiding methods based on audio and video synchronization in MP4 container","authors":"Imdad MaungMaung, Koksheik Wong, Kiyoshi Tanaka","doi":"10.1109/ISPACS.2016.7824699","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824699","url":null,"abstract":"In this work, two reversible data hiding methods are proposed by exploiting the audio-video synchronization process in the MP4 container. Specifically, audio-video synchronization information is stored as time-to-sample information in the stts box of the MP4 container. In Method 1, the number of sample counts in the stts box is decomposed into multiple integers, where each integer represents the decimal equivalent of the payload (represented in binary) to be hidden. Method 1 completely preserves the audio-video synchronization before and after hiding data into the MP4 container. To suppress bitstream size increment and improve the number of bits that can be hidden, Method 2 is proposed by manipulating both timescale and duration of sample δ to hide data. Basic performance of the proposed methods are verified through experiments on various H.264/AVC and AAC compressed short movie clips downloaded from YouTube. In the worst case scenario, a negligible bitstream size increment of < 0.0447% and absolute synchronization error of < 13 milliseconds were observed when payload of length 1024 bits is embedded into the movie clips.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":"92 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122577696","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 : 2016-10-01DOI: 10.1109/ISPACS.2016.7824733
Thanat Nonthaputha, M. Kumngern, P. Moungnoul
This paper presents a new digital-to-analog (D/A) converter circuit using second-generation current conveyors (CCIIs) which works as analogue switches. In this case, CCII is worked as current conveyor analogue switch (CCAS). A bit stream of digital code is used to control the switch for obtaining the current level at the output terminal. Then, bits D/A converter can be obtained simply by parallel connecting of CCASs. The proposed structure is suitable for low power supply voltage. The simulation results are used to confirm the workability of the proposed structure.
{"title":"CMOS D/A converter using current conveyor analogue switches","authors":"Thanat Nonthaputha, M. Kumngern, P. Moungnoul","doi":"10.1109/ISPACS.2016.7824733","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824733","url":null,"abstract":"This paper presents a new digital-to-analog (D/A) converter circuit using second-generation current conveyors (CCIIs) which works as analogue switches. In this case, CCII is worked as current conveyor analogue switch (CCAS). A bit stream of digital code is used to control the switch for obtaining the current level at the output terminal. Then, bits D/A converter can be obtained simply by parallel connecting of CCASs. The proposed structure is suitable for low power supply voltage. The simulation results are used to confirm the workability of the proposed structure.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125439492","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 : 2016-10-01DOI: 10.1109/ISPACS.2016.7824687
K. Inoue, K. Hara, K. Urahama
We propose two methods for recovering the reflectance spectra of given colorimetric data by using the nonnegative constraints in reflectance spectra. We formulate the problem of reflectance spectra recovery as a non-negative least squares problem and solve it with two iterative methods. Experimental results demonstrate that the two methods give similar recovery results, where Macbeth ColorChecker data are used for recovering the reflectance spectra of Neugebauer primary colors. We also transform the recovered reflectance spectra into tristimulus values to visualize them, where an ad hoc scaling operation is introduced for brightening the recovered colors.
{"title":"Reflectance spectra recovery with non-negativity constraints","authors":"K. Inoue, K. Hara, K. Urahama","doi":"10.1109/ISPACS.2016.7824687","DOIUrl":"https://doi.org/10.1109/ISPACS.2016.7824687","url":null,"abstract":"We propose two methods for recovering the reflectance spectra of given colorimetric data by using the nonnegative constraints in reflectance spectra. We formulate the problem of reflectance spectra recovery as a non-negative least squares problem and solve it with two iterative methods. Experimental results demonstrate that the two methods give similar recovery results, where Macbeth ColorChecker data are used for recovering the reflectance spectra of Neugebauer primary colors. We also transform the recovered reflectance spectra into tristimulus values to visualize them, where an ad hoc scaling operation is introduced for brightening the recovered colors.","PeriodicalId":131543,"journal":{"name":"2016 International Symposium on Intelligent Signal Processing and Communication Systems (ISPACS)","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-10-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127788096","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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