Pub Date : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577107
E. Aboutanios, B. Mulgrew
The detection of signals with known templates embedded in zero-mean coloured Gaussian interference is relevant to many fields such as radar, sonar, seismology and biomedicine to name a few. Traditional detection algorithms, such as the GLRT and AMF, require a training data set. Recently, single data set (SDS) algorithms, namely the GMLED and MLED, have been proposed to deal with the case where training data may not be available. In this paper, we examine the performance of these algorithms under template (or steering vector) mismatch. We identify three types of mismatch, namely the spatial steering vector mismatch, temporal steering vector mismatch and mismatch in both steering vectors. In each mismath case we derive the expected signal to noise ratio loss with respect to the corresponding matched case. Simulation results are given which show that the SDS algorithms are more sensitive to mismatch mainly due to the interaction between the signal and subspaces estimation. However, this increased sensitivity to mismatch is closely related to the ability to resolve close signals. Therefore, the SDS algorithms exhibit higher resolution
{"title":"Assessment of the single data set detection algorithms under template mismatch","authors":"E. Aboutanios, B. Mulgrew","doi":"10.1109/ISSPIT.2005.1577107","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577107","url":null,"abstract":"The detection of signals with known templates embedded in zero-mean coloured Gaussian interference is relevant to many fields such as radar, sonar, seismology and biomedicine to name a few. Traditional detection algorithms, such as the GLRT and AMF, require a training data set. Recently, single data set (SDS) algorithms, namely the GMLED and MLED, have been proposed to deal with the case where training data may not be available. In this paper, we examine the performance of these algorithms under template (or steering vector) mismatch. We identify three types of mismatch, namely the spatial steering vector mismatch, temporal steering vector mismatch and mismatch in both steering vectors. In each mismath case we derive the expected signal to noise ratio loss with respect to the corresponding matched case. Simulation results are given which show that the SDS algorithms are more sensitive to mismatch mainly due to the interaction between the signal and subspaces estimation. However, this increased sensitivity to mismatch is closely related to the ability to resolve close signals. Therefore, the SDS algorithms exhibit higher resolution","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"27 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121548773","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 : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577102
D. Serpanos, M. Gamvrili
Scheduling algorithms for virtual output queuing (VOQ) switches aim to provide high performance (low latency and high throughput), fairness as well as easy and low cost implementation. These goals have led to the development of a wide range of schedulers that are characterized by deterministic decision making. Randomness in decision making enables high performance scheduling as well as fair treatment of requests, when used appropriately. However, it has been proven to be costly in switching environments, which require high speed decisions, as the experience with the first randomized algorithm, namely PIM, is shown. In this paper, we introduce a class of practical scheduling algorithms that employ randomness in a fashion that leads to fast, low cost implementations. The class of algorithms, named random on-line matching (ROLM), employs randomness effectively, calculating maximal matchings and thus, leading to high performance schedulers with optimal mean service time that outperform significantly the competitive deterministic schedulers; furthermore, it employs randomness efficiently, resulting to schedulers that can accommodate effectively high-speed links
{"title":"Randomized on-line matching (ROLM): randomized scheduling algorithms for high-speed cell-based switches","authors":"D. Serpanos, M. Gamvrili","doi":"10.1109/ISSPIT.2005.1577102","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577102","url":null,"abstract":"Scheduling algorithms for virtual output queuing (VOQ) switches aim to provide high performance (low latency and high throughput), fairness as well as easy and low cost implementation. These goals have led to the development of a wide range of schedulers that are characterized by deterministic decision making. Randomness in decision making enables high performance scheduling as well as fair treatment of requests, when used appropriately. However, it has been proven to be costly in switching environments, which require high speed decisions, as the experience with the first randomized algorithm, namely PIM, is shown. In this paper, we introduce a class of practical scheduling algorithms that employ randomness in a fashion that leads to fast, low cost implementations. The class of algorithms, named random on-line matching (ROLM), employs randomness effectively, calculating maximal matchings and thus, leading to high performance schedulers with optimal mean service time that outperform significantly the competitive deterministic schedulers; furthermore, it employs randomness efficiently, resulting to schedulers that can accommodate effectively high-speed links","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"60 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121834159","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 : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577215
N. Myridis
In this lecture we investigate the Hough transform (HuT) on geometrical (non-probabilistic) distributions, i.e. on loci of point set distributions in the spatial space. Such distributions are assumed to be linear, circular, spiral etc. The usefulness of the assumption of such distributions is multiple according to the existence of them in a huge variety of image processing problems. Furthermore we present innovative elements of HuT which may lead to the development of means for evaluating HuT of complicated geometrical figures
{"title":"The Hough transform (HuT) on geometrical distributions (geodistributions)","authors":"N. Myridis","doi":"10.1109/ISSPIT.2005.1577215","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577215","url":null,"abstract":"In this lecture we investigate the Hough transform (HuT) on geometrical (non-probabilistic) distributions, i.e. on loci of point set distributions in the spatial space. Such distributions are assumed to be linear, circular, spiral etc. The usefulness of the assumption of such distributions is multiple according to the existence of them in a huge variety of image processing problems. Furthermore we present innovative elements of HuT which may lead to the development of means for evaluating HuT of complicated geometrical figures","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"2650 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115030955","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 : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577110
G. Metikas, S. Olhede
We consider characterisation and analysis of images with dominant local frequencies: this class contains two-dimensional signals written as sums of amplitude and frequency modulated components. We introduce the usage of the multiple vector-valued continuous Morse wavelets for analysis of such signals. We calculate the continuous wavelet transform and the associated scalogram of an image to extract its local amplitude, frequency and dominant orientation. For images contaminated by additive noise, we estimate the local features of the image using several orthogonal mother wavelet functions with optimal localisation: this allows for the construction of estimators of local image features with reduced variability and bias
{"title":"Multiple wavelet analysis of amplitude/frequency modulated images","authors":"G. Metikas, S. Olhede","doi":"10.1109/ISSPIT.2005.1577110","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577110","url":null,"abstract":"We consider characterisation and analysis of images with dominant local frequencies: this class contains two-dimensional signals written as sums of amplitude and frequency modulated components. We introduce the usage of the multiple vector-valued continuous Morse wavelets for analysis of such signals. We calculate the continuous wavelet transform and the associated scalogram of an image to extract its local amplitude, frequency and dominant orientation. For images contaminated by additive noise, we estimate the local features of the image using several orthogonal mother wavelet functions with optimal localisation: this allows for the construction of estimators of local image features with reduced variability and bias","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128037909","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 : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577105
Jingying Chen, Bernard Paul Tiddeman
Head pose tracking is very important in many human machine interfaces. In this paper, a stereo head pose tracking system is proposed, which detects and tracks six facial features (pupils, nostrils and lip corners) automatically when a human face appears in front of two low-cost USB cameras. The epipolar constraint and a 3D facial feature model are used to improve the accuracy and robustness of the system. This system has the advantages of automatically detecting the facial features and recovering the features lost during the tracking process. Encouraging results have been obtained using the proposed system
{"title":"A real-time stereo head pose tracking system","authors":"Jingying Chen, Bernard Paul Tiddeman","doi":"10.1109/ISSPIT.2005.1577105","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577105","url":null,"abstract":"Head pose tracking is very important in many human machine interfaces. In this paper, a stereo head pose tracking system is proposed, which detects and tracks six facial features (pupils, nostrils and lip corners) automatically when a human face appears in front of two low-cost USB cameras. The epipolar constraint and a 3D facial feature model are used to improve the accuracy and robustness of the system. This system has the advantages of automatically detecting the facial features and recovering the features lost during the tracking process. Encouraging results have been obtained using the proposed system","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"31 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131351711","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 : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577138
Zhijun Zhao, R. Blahut
Both blind and nonblind demodulation algorithms based on the Richardson-Lucy algorithm for the one-dimensional intersymbol interference channel with nonnegative impulse response function are proposed and compared numerically with the Viterbi demodulation algorithm and the demodulation algorithms based on the Wiener filter. Simulation results show that the demodulation algorithms based on the Richardson-Lucy algorithm have better bit error rate performance than the demodulation algorithms based on the Wiener filter. With the progressive thresholding, the bit error rate curves of both blind and nonblind Richardson-Lucy-based algorithms are within 0.3 dB from that of the Viterbi algorithm throughout the entire simulated signal-to-noise ratio range. While the complexity of the Viterbi algorithm depends critically on the length of the channel impulse response function, the complexity of the demodulation algorithms based on the Richardson-Lucy algorithm only weakly depends on it
{"title":"Blind and nonblind nonnegative impulse response ISI channel demodulation using the Richardson-Lucy algorithm","authors":"Zhijun Zhao, R. Blahut","doi":"10.1109/ISSPIT.2005.1577138","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577138","url":null,"abstract":"Both blind and nonblind demodulation algorithms based on the Richardson-Lucy algorithm for the one-dimensional intersymbol interference channel with nonnegative impulse response function are proposed and compared numerically with the Viterbi demodulation algorithm and the demodulation algorithms based on the Wiener filter. Simulation results show that the demodulation algorithms based on the Richardson-Lucy algorithm have better bit error rate performance than the demodulation algorithms based on the Wiener filter. With the progressive thresholding, the bit error rate curves of both blind and nonblind Richardson-Lucy-based algorithms are within 0.3 dB from that of the Viterbi algorithm throughout the entire simulated signal-to-noise ratio range. While the complexity of the Viterbi algorithm depends critically on the length of the channel impulse response function, the complexity of the demodulation algorithms based on the Richardson-Lucy algorithm only weakly depends on it","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130596171","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 : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577179
E. Abdel-Raheem
This paper investigates computationally-efficient adaptive algorithms suitable for echo cancellation applications. Power-of-two (PO2) quantization for the error and/or regressor vector is added to a selective-partial-coefficient-update algorithm (SPCUA) to further reduce the amount of computations. A simple stop-and-go procedure is included to save more computations when the generated error magnitude is below a certain limit. Simulation results with comparisons are included to show the advantages of the improved algorithms
{"title":"On computationally-efficient NLMS-based algorithms for echo cancellation applications","authors":"E. Abdel-Raheem","doi":"10.1109/ISSPIT.2005.1577179","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577179","url":null,"abstract":"This paper investigates computationally-efficient adaptive algorithms suitable for echo cancellation applications. Power-of-two (PO2) quantization for the error and/or regressor vector is added to a selective-partial-coefficient-update algorithm (SPCUA) to further reduce the amount of computations. A simple stop-and-go procedure is included to save more computations when the generated error magnitude is below a certain limit. Simulation results with comparisons are included to show the advantages of the improved algorithms","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"86 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130567150","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 : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577080
T. Fujino, J. Nakamura
Recently we have proposed a multi-carrier code division multiple access (MC-CDMA) scheme for the downlink of mobile communications to combat the channel blocking in flat fading multi-input single-output (MISO) channels. In this paper, the previous proposal is extended to the case of frequency-selective fading multi-input multi-output (MIMO) channels. An enhanced scheme is also proposed to obtain better bit error rate (BER) performance and to facilitate signal detection. BER performance of the proposed system using different detection methods is evaluated by numerical calculation. Without decreasing time, frequency, and transmit (Tx) power efficiencies, the proposed system allows us to extract all the Tx data with relatively good BER even when channel blocking is encountered
{"title":"A multicarrier CDMA system for combating channel blocking in frequency-selective fading MIMO channel","authors":"T. Fujino, J. Nakamura","doi":"10.1109/ISSPIT.2005.1577080","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577080","url":null,"abstract":"Recently we have proposed a multi-carrier code division multiple access (MC-CDMA) scheme for the downlink of mobile communications to combat the channel blocking in flat fading multi-input single-output (MISO) channels. In this paper, the previous proposal is extended to the case of frequency-selective fading multi-input multi-output (MIMO) channels. An enhanced scheme is also proposed to obtain better bit error rate (BER) performance and to facilitate signal detection. BER performance of the proposed system using different detection methods is evaluated by numerical calculation. Without decreasing time, frequency, and transmit (Tx) power efficiencies, the proposed system allows us to extract all the Tx data with relatively good BER even when channel blocking is encountered","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132077379","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 : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577087
H. Belkacemi, M. Djeddi, S. Marcos
We consider the problem of joint angle and Doppler estimation for space-time adaptive processing (STAP) airborne radar in non Gaussian clutter modeled as a complex symmetric alpha stable (SaS) process. We introduce a new robust MUSIC algorithm named PFLOM-MUSIC which is obtained using a new class of covariance matrix estimate based on the phased fractional lower-order moment(PFLOM). Such matrix and the subspaces spanned by its eigenvectors serve as a basis for a high-resolution angle/Doppler estimation method. The performance of the PFLOM-MUSIC is examined by comparing the estimation results with those of the sign covariance matrix estimate SCM-MUSIC (T. Liu and J. M. Mendel, Aug. 2001). Extensive Simulations demonstrate that our proposed algorithm outperforms the previously reported estimators
将空时自适应处理(STAP)机载雷达在非高斯杂波环境中的联合角和多普勒估计问题建模为复杂对称α稳定过程。提出了一种新的基于相位分数阶矩(PFLOM)的协方差矩阵估计的鲁棒MUSIC算法PFLOM-MUSIC。该矩阵及其特征向量张成的子空间是高分辨率角度/多普勒估计方法的基础。通过将估计结果与符号协方差矩阵估计SCM-MUSIC的估计结果进行比较，检验了PFLOM-MUSIC的性能(T. Liu and J. M. Mendel, august 2001)。大量的仿真表明，我们提出的算法优于先前报道的估计器
{"title":"Robust subspace technique for joint angle/Doppler estimation in /spl alpha/-stable clutter using phased fractional lower-order moments","authors":"H. Belkacemi, M. Djeddi, S. Marcos","doi":"10.1109/ISSPIT.2005.1577087","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577087","url":null,"abstract":"We consider the problem of joint angle and Doppler estimation for space-time adaptive processing (STAP) airborne radar in non Gaussian clutter modeled as a complex symmetric alpha stable (SaS) process. We introduce a new robust MUSIC algorithm named PFLOM-MUSIC which is obtained using a new class of covariance matrix estimate based on the phased fractional lower-order moment(PFLOM). Such matrix and the subspaces spanned by its eigenvectors serve as a basis for a high-resolution angle/Doppler estimation method. The performance of the PFLOM-MUSIC is examined by comparing the estimation results with those of the sign covariance matrix estimate SCM-MUSIC (T. Liu and J. M. Mendel, Aug. 2001). Extensive Simulations demonstrate that our proposed algorithm outperforms the previously reported estimators","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133869086","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 : 2005-12-21DOI: 10.1109/ISSPIT.2005.1577068
L. Golipour, S. Gazor
This paper presents a new approach to an auditory model which matches closely the response patterns in physiological data for single tone inputs. Including the biophysical complexity of the wave motion in the cochlea and considering all terms of the motion equation, the new model can evaluate the auditory spectrum using the Basilar membrane (BM) displacement and the inner hair cells' (IHCs) firing rate for all input signals, including speech. We employ the partial differential motion equations of the BM and its parameters measured for the human auditory system, and design an algorithm which uses short time Fourier transform (STFT) to compute the output for speech stimulus. The idea is to isolate the input signal in the vicinity of a time-window and try to follow the changes in its frequencies and their influences on the signal perceived by the auditory system. The new model includes the nonlinearity action of outer hair cells (OHCs) and provides a new auditory spectrum for speech inputs in the real time domain which reflects a proper view of propagating signal in the cochlea. Despite most of the previous models this model can track the effects of high formant frequencies in the human cochlea as well. This model is a new signal processing tool for studying the response of the auditory system to transient signals which is highly demanded in various speech enhancement and audio coding algorithms
{"title":"A biophysical model of the human cochlea for speech stimulus using STFT","authors":"L. Golipour, S. Gazor","doi":"10.1109/ISSPIT.2005.1577068","DOIUrl":"https://doi.org/10.1109/ISSPIT.2005.1577068","url":null,"abstract":"This paper presents a new approach to an auditory model which matches closely the response patterns in physiological data for single tone inputs. Including the biophysical complexity of the wave motion in the cochlea and considering all terms of the motion equation, the new model can evaluate the auditory spectrum using the Basilar membrane (BM) displacement and the inner hair cells' (IHCs) firing rate for all input signals, including speech. We employ the partial differential motion equations of the BM and its parameters measured for the human auditory system, and design an algorithm which uses short time Fourier transform (STFT) to compute the output for speech stimulus. The idea is to isolate the input signal in the vicinity of a time-window and try to follow the changes in its frequencies and their influences on the signal perceived by the auditory system. The new model includes the nonlinearity action of outer hair cells (OHCs) and provides a new auditory spectrum for speech inputs in the real time domain which reflects a proper view of propagating signal in the cochlea. Despite most of the previous models this model can track the effects of high formant frequencies in the human cochlea as well. This model is a new signal processing tool for studying the response of the auditory system to transient signals which is highly demanded in various speech enhancement and audio coding algorithms","PeriodicalId":421826,"journal":{"name":"Proceedings of the Fifth IEEE International Symposium on Signal Processing and Information Technology, 2005.","volume":"141 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2005-12-21","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133618042","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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