Pub Date : 2014-12-01DOI: 10.1109/APSIPA.2014.7041611
Fan Chen, Yao Qin, Hongjie He
To improve the ability against different counterfeiting attacks, a watermarking algorithm is proposed for binary-text image (BTI) authentication. To protect the uniform regions in BTI, the watermark information of a fixed-size block is generated according to the content of it and divided into three parts. One part is embedded in the flippable pixels of itself, and the other two parts are respectively embedded in the flippable pixels of other two blocks in BTI, which are randomly chosen based on the secret key. This strategy can not only introduce the block-wise dependence, but also make it possible for the authentication watermark of a uniform block to be embedded in the BTI. In the tamper detection stage, a multi-block based statistic detection method is designed to verify the validity of an image block. Simulation results show that the proposed algorithm can achieve a good imperceptibility and have an ability resisting the maliciously attacks such as collage attack, delete tampering, replace tampering etc.
{"title":"Multi-block dependency based watermarking scheme for binary-text image authentication","authors":"Fan Chen, Yao Qin, Hongjie He","doi":"10.1109/APSIPA.2014.7041611","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041611","url":null,"abstract":"To improve the ability against different counterfeiting attacks, a watermarking algorithm is proposed for binary-text image (BTI) authentication. To protect the uniform regions in BTI, the watermark information of a fixed-size block is generated according to the content of it and divided into three parts. One part is embedded in the flippable pixels of itself, and the other two parts are respectively embedded in the flippable pixels of other two blocks in BTI, which are randomly chosen based on the secret key. This strategy can not only introduce the block-wise dependence, but also make it possible for the authentication watermark of a uniform block to be embedded in the BTI. In the tamper detection stage, a multi-block based statistic detection method is designed to verify the validity of an image block. Simulation results show that the proposed algorithm can achieve a good imperceptibility and have an ability resisting the maliciously attacks such as collage attack, delete tampering, replace tampering etc.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"67 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125025935","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 : 2014-12-01DOI: 10.1109/APSIPA.2014.7041690
Withawat Tangtrongpairoj, T. Higashino, M. Okada
Radio over Fiber (RoF) is a promising solution for wireless access services by transferring the heterogeneous radio signal via the optical fiber link. However, RoF devices have nonlinear characteristics which create intermodulation products in system. The intermodulation distortion (IMD) interferes uplink RF signals in the presence of coupling between downlink and uplink antennas in the base station (BS). This paper proposed the performance evaluation due to coupled downlink interfere to uplink antenna. The carrier to distortion plus noise ratio (CDNR) is evaluated for all combinations. By using NS3 network simulator, the result shows the best combination achieves better performance. Which coupled downlink interfere in uplink signal can be reduced when amount of downlink packet is decreased.
{"title":"Quality-based channel selection in multi-channel radio-over-fiber system","authors":"Withawat Tangtrongpairoj, T. Higashino, M. Okada","doi":"10.1109/APSIPA.2014.7041690","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041690","url":null,"abstract":"Radio over Fiber (RoF) is a promising solution for wireless access services by transferring the heterogeneous radio signal via the optical fiber link. However, RoF devices have nonlinear characteristics which create intermodulation products in system. The intermodulation distortion (IMD) interferes uplink RF signals in the presence of coupling between downlink and uplink antennas in the base station (BS). This paper proposed the performance evaluation due to coupled downlink interfere to uplink antenna. The carrier to distortion plus noise ratio (CDNR) is evaluated for all combinations. By using NS3 network simulator, the result shows the best combination achieves better performance. Which coupled downlink interfere in uplink signal can be reduced when amount of downlink packet is decreased.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"73 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122797541","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 : 2014-12-01DOI: 10.1109/APSIPA.2014.7041567
M. Mimura, Tatsuya Kawahara
Unsupervised speaker adaptation of Deep Neural Network (DNN) is investigated for lecture transcription tasks, in which a single speaker gives a long speech and thus speaker adaptation is important. The proposed method selects similar speakers to the test data (test speaker) from the training database, which are used for retraining the baseline DNN. Several speaker characteristic features are defined for the speaker similarity measure. The feature based on Universal Background Model (UBM) and principal component analysis (PCA) achieves the best performance, resulting in a significant improvement from the baseline DNN and also from the adapted GMM-HMM system. The method is combined with a naive adaptation method using the initial ASR hypothesis of the test data, and an additional improvement is achieved.
{"title":"Unsupervised speaker adaptation of DNN-HMM by selecting similar speakers for lecture transcription","authors":"M. Mimura, Tatsuya Kawahara","doi":"10.1109/APSIPA.2014.7041567","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041567","url":null,"abstract":"Unsupervised speaker adaptation of Deep Neural Network (DNN) is investigated for lecture transcription tasks, in which a single speaker gives a long speech and thus speaker adaptation is important. The proposed method selects similar speakers to the test data (test speaker) from the training database, which are used for retraining the baseline DNN. Several speaker characteristic features are defined for the speaker similarity measure. The feature based on Universal Background Model (UBM) and principal component analysis (PCA) achieves the best performance, resulting in a significant improvement from the baseline DNN and also from the adapted GMM-HMM system. The method is combined with a naive adaptation method using the initial ASR hypothesis of the test data, and an additional improvement is achieved.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"182 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125828469","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 : 2014-12-01DOI: 10.1109/APSIPA.2014.7041537
Yanping Zhou, Y. Duan, Jun Sun, Zongming Guo
This paper focuses on the intra frame rate control of H.264/AVC High Profile and introduces a new frame gradient-based rate control algorithm. In this algorithm, a rate-gradient-quantization parameter model with frame gradient employed as frame complexity is proposed. Then, a proportional feedback scheme, along with an adaptive optimization method, is presented to achieve constant bitrate. Rigorous experiments covering various sequences of different target rates are carried out. Experimental results show that the proposed rate control method outperforms JM16.0 by offering a more constant rate output and reducing rate fluctuation, without video quality loss.
本文重点研究了H.264/AVC High Profile的帧内速率控制，提出了一种基于帧梯度的帧内速率控制算法。在该算法中，提出了一种以帧梯度作为帧复杂度的速率梯度量化参数模型。然后，提出了一种比例反馈方案，并结合自适应优化方法来实现恒定比特率。对不同目标速率的各种序列进行了严格的实验。实验结果表明，该方法在不影响视频质量的前提下，提供了更稳定的速率输出，减少了速率波动，优于JM16.0。
{"title":"Proportional feedback based rate control for intra frame of H.264/AVC high profile","authors":"Yanping Zhou, Y. Duan, Jun Sun, Zongming Guo","doi":"10.1109/APSIPA.2014.7041537","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041537","url":null,"abstract":"This paper focuses on the intra frame rate control of H.264/AVC High Profile and introduces a new frame gradient-based rate control algorithm. In this algorithm, a rate-gradient-quantization parameter model with frame gradient employed as frame complexity is proposed. Then, a proportional feedback scheme, along with an adaptive optimization method, is presented to achieve constant bitrate. Rigorous experiments covering various sequences of different target rates are carried out. Experimental results show that the proposed rate control method outperforms JM16.0 by offering a more constant rate output and reducing rate fluctuation, without video quality loss.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"18 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127201372","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 : 2014-12-01DOI: 10.1109/APSIPA.2014.7041644
Etkin Baris Ozgul, Somchaya Liemhetcharat, K. H. Low
Multi-agent research has focused on finding the optimal team for a task. Many approaches assume that the performance of the agents are known a priori. We are interested in ad hoc teams, where the agents' algorithms and performance are initially unknown. We focus on the task of modeling the performance of single agents through observation in training environments, and using the learned models to partition a new environment for a multi-agent team. The goal is to minimize the number of agents used, while maintaining a performance threshold of the multi-agent team. We contribute a novel model to learn the agent's performance through observations, and a partitioning algorithm that minimizes the team size. We evaluate our algorithms in simulation, and show the efficacy of our learn model and partitioning algorithm.
{"title":"Multi-agent ad hoc team partitioning by observing and modeling single-agent performance","authors":"Etkin Baris Ozgul, Somchaya Liemhetcharat, K. H. Low","doi":"10.1109/APSIPA.2014.7041644","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041644","url":null,"abstract":"Multi-agent research has focused on finding the optimal team for a task. Many approaches assume that the performance of the agents are known a priori. We are interested in ad hoc teams, where the agents' algorithms and performance are initially unknown. We focus on the task of modeling the performance of single agents through observation in training environments, and using the learned models to partition a new environment for a multi-agent team. The goal is to minimize the number of agents used, while maintaining a performance threshold of the multi-agent team. We contribute a novel model to learn the agent's performance through observations, and a partitioning algorithm that minimizes the team size. We evaluate our algorithms in simulation, and show the efficacy of our learn model and partitioning algorithm.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"20 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114247026","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 : 2014-12-01DOI: 10.1109/APSIPA.2014.7041624
Jia-Ching Wang, Chang-Hong Lin, En-Ting Chen, P. Chang
This paper aims to propose a new set of acoustic features based on spectral-temporal receptive fields (STRFs). The STRF is an analysis method for studying physiological model of the mammalian auditory system in spectral-temporal domain. It has two different parts: one is the rate (in Hz) which represents the temporal response and the other is the scale (in cycle/octave) which represents the spectral response. With the obtained STRF, we propose an effective acoustic feature. First, the energy of each scale is calculated from the STRF. The logarithmic operation is then imposed on the scale energies. Finally, the discrete Cosine transform is applied to generate the proposed STRF feature. In our experiments, we combine the proposed STRF feature with conventional Mel frequency cepstral coefficients (MFCCs) to verify its effectiveness. In a noise-free environment, the proposed feature can increase the recognition rate by 17.48%. Moreover, the increase in the recognition rate ranges from 5% to 12% in noisy environments.
{"title":"Spectral-temporal receptive fields and MFCC balanced feature extraction for noisy speech recognition","authors":"Jia-Ching Wang, Chang-Hong Lin, En-Ting Chen, P. Chang","doi":"10.1109/APSIPA.2014.7041624","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041624","url":null,"abstract":"This paper aims to propose a new set of acoustic features based on spectral-temporal receptive fields (STRFs). The STRF is an analysis method for studying physiological model of the mammalian auditory system in spectral-temporal domain. It has two different parts: one is the rate (in Hz) which represents the temporal response and the other is the scale (in cycle/octave) which represents the spectral response. With the obtained STRF, we propose an effective acoustic feature. First, the energy of each scale is calculated from the STRF. The logarithmic operation is then imposed on the scale energies. Finally, the discrete Cosine transform is applied to generate the proposed STRF feature. In our experiments, we combine the proposed STRF feature with conventional Mel frequency cepstral coefficients (MFCCs) to verify its effectiveness. In a noise-free environment, the proposed feature can increase the recognition rate by 17.48%. Moreover, the increase in the recognition rate ranges from 5% to 12% in noisy environments.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114445257","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 : 2014-12-01DOI: 10.1109/APSIPA.2014.7041689
Jinwei Sun, Fei Ma, Boyan Huang, Liang Wen
Narrowband active noise control (ANC) systems enjoy good performance where sinusoidal signals dominate in the primary noise, on condition that a reference signal of the same frequencies with the primary noise is given. However, frequencies of the reference signal provided by nonacoustic sensors are usually different from that of the primary noise due to temperature changes, aging, etc. Such frequency mismatch (FM) will make the narrowband ANC systems unable to suppress the primary noise effectively, even render them useless. In this paper, we propose a new narrowband ANC system that integrated with a frequency estimation subsystem. The frequency estimation is obtained from a spectrum computation based on an adaptive linear prediction filter. The estimated frequencies are used by the cosine signal generator to produce a more accurate reference signal to the main controller, thus the performance deterioration caused by FM can be mitigated. The effectiveness of the proposed system has been confirmed by numerous simulations.
{"title":"A narrowband active noise control system with frequency mismatch compensation","authors":"Jinwei Sun, Fei Ma, Boyan Huang, Liang Wen","doi":"10.1109/APSIPA.2014.7041689","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041689","url":null,"abstract":"Narrowband active noise control (ANC) systems enjoy good performance where sinusoidal signals dominate in the primary noise, on condition that a reference signal of the same frequencies with the primary noise is given. However, frequencies of the reference signal provided by nonacoustic sensors are usually different from that of the primary noise due to temperature changes, aging, etc. Such frequency mismatch (FM) will make the narrowband ANC systems unable to suppress the primary noise effectively, even render them useless. In this paper, we propose a new narrowband ANC system that integrated with a frequency estimation subsystem. The frequency estimation is obtained from a spectrum computation based on an adaptive linear prediction filter. The estimated frequencies are used by the cosine signal generator to produce a more accurate reference signal to the main controller, thus the performance deterioration caused by FM can be mitigated. The effectiveness of the proposed system has been confirmed by numerous simulations.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"21 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126276962","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 : 2014-12-01DOI: 10.1109/APSIPA.2014.7041638
Fuming Fang, T. Shinozaki, Takao Kobayashi
A switch-to-speech interface can provide a means of interactive communication as a support system for people with disabilities with voluntary movements. Any motion of a part of the body, such as eye movements, can be used for the switch input. The number of possible switch operations varies from person to person, but the bandwidth is generally quite limited. Therefore, efficient input protocols are needed to map the switch operations to pronunciations. Meanwhile, the protocol must be easily learnable so that anyone can use it. To this end, we propose a protocol recommendation method that can accept individual requirements in switch operations. This method suggests a customized protocol for each user of the interface that is both speedy to enter and easy to remember. The two main ideas in the protocol design are utilizing the knowledge about the alphabet table that everyone already knows and improving the input speed and learnability by allowing ambiguity in the switch to pronunciation conversion. The conversion errors due to the ambiguity are offset by an N-gram language model. The performance of the protocols was evaluated through simulations and the measured values obtained from research participants, and the advantage of the proposed method is shown.
{"title":"An automatic input protocol recommendation method for tailored switch-to-speech communication aid systems","authors":"Fuming Fang, T. Shinozaki, Takao Kobayashi","doi":"10.1109/APSIPA.2014.7041638","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041638","url":null,"abstract":"A switch-to-speech interface can provide a means of interactive communication as a support system for people with disabilities with voluntary movements. Any motion of a part of the body, such as eye movements, can be used for the switch input. The number of possible switch operations varies from person to person, but the bandwidth is generally quite limited. Therefore, efficient input protocols are needed to map the switch operations to pronunciations. Meanwhile, the protocol must be easily learnable so that anyone can use it. To this end, we propose a protocol recommendation method that can accept individual requirements in switch operations. This method suggests a customized protocol for each user of the interface that is both speedy to enter and easy to remember. The two main ideas in the protocol design are utilizing the knowledge about the alphabet table that everyone already knows and improving the input speed and learnability by allowing ambiguity in the switch to pronunciation conversion. The conversion errors due to the ambiguity are offset by an N-gram language model. The performance of the protocols was evaluated through simulations and the measured values obtained from research participants, and the advantage of the proposed method is shown.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"78 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126306398","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 : 2014-12-01DOI: 10.1109/APSIPA.2014.7041814
P. Temdee, Wacharawan Intayoad
Web based learning has been promoting alternative way of learning for decades. The difficulty of web based learning is to provide the appropriate support for the learners so that the learners will not get lost and their learning achievements can be ensured. This paper thus proposes the method for discovering learning patterns of the learners on web based learning particularly for ensuring the learning achievement. The learning pattern is discovered by analyzing the interactions among the learners and the learning objects with social network analysis. Then, the achievement learning pattern is finally determined by analyzing the sets of obtained social network measurements. The interaction data is gathered from online course named introduction to Information Technology in the 2013 academic year, particularly for spreadsheet content module having 10 learning objects. The interaction patterns only of two groups of students including scientific and nonscientific background knowledge who pass the spreadsheet examination are analyzed. Finally, learning patterns ensuring learning achievement for spreadsheet content module of those students having different background knowledge is revealed.
{"title":"Discovering and analyzing learning pattern on web based learning using social network analysis","authors":"P. Temdee, Wacharawan Intayoad","doi":"10.1109/APSIPA.2014.7041814","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041814","url":null,"abstract":"Web based learning has been promoting alternative way of learning for decades. The difficulty of web based learning is to provide the appropriate support for the learners so that the learners will not get lost and their learning achievements can be ensured. This paper thus proposes the method for discovering learning patterns of the learners on web based learning particularly for ensuring the learning achievement. The learning pattern is discovered by analyzing the interactions among the learners and the learning objects with social network analysis. Then, the achievement learning pattern is finally determined by analyzing the sets of obtained social network measurements. The interaction data is gathered from online course named introduction to Information Technology in the 2013 academic year, particularly for spreadsheet content module having 10 learning objects. The interaction patterns only of two groups of students including scientific and nonscientific background knowledge who pass the spreadsheet examination are analyzed. Finally, learning patterns ensuring learning achievement for spreadsheet content module of those students having different background knowledge is revealed.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"145 5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129695065","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 : 2014-12-01DOI: 10.1109/APSIPA.2014.7041527
Ling-song Zhou, C. Bao, Mao-shen Jia, Bing Bu
3D audio technology based on binaural reproduction requires the Head-Related Transfer Function (HRTF) datasets to be available for all possible distance. However, due to the tedious work of measurement and large volume of resulting datasets, the HRTF is typically measured only for sources located at a fixed distance. In this paper, the concept of virtual loudspeaker arrays is utilized to achieve range extrapolation of the measured HRTF datasets at a single range. The virtual loudspeaker is driven by Higher Order Ambisonics (HOA). Specially, to restrict the near-field effect of HOA, a compensation method of modified Wiener filter is proposed. The simulation results indicate that the proposed method provides effective range extrapolation of HRTF.
{"title":"Range extrapolation of Head-Related Transfer Function using improved Higher Order Ambisonics","authors":"Ling-song Zhou, C. Bao, Mao-shen Jia, Bing Bu","doi":"10.1109/APSIPA.2014.7041527","DOIUrl":"https://doi.org/10.1109/APSIPA.2014.7041527","url":null,"abstract":"3D audio technology based on binaural reproduction requires the Head-Related Transfer Function (HRTF) datasets to be available for all possible distance. However, due to the tedious work of measurement and large volume of resulting datasets, the HRTF is typically measured only for sources located at a fixed distance. In this paper, the concept of virtual loudspeaker arrays is utilized to achieve range extrapolation of the measured HRTF datasets at a single range. The virtual loudspeaker is driven by Higher Order Ambisonics (HOA). Specially, to restrict the near-field effect of HOA, a compensation method of modified Wiener filter is proposed. The simulation results indicate that the proposed method provides effective range extrapolation of HRTF.","PeriodicalId":231382,"journal":{"name":"Signal and Information Processing Association Annual Summit and Conference (APSIPA), 2014 Asia-Pacific","volume":"14 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2014-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127604411","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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