Browsing by Author "Chang, Wen-Whei"

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Biometric Identification Using JPEG2000 Compressed ECG Signals
(2015-10-01)
Wu, Hung-Tsai
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Wu, Yi-Ting
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Chang, Wen-Whei
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電信工程研究所
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Institute of Communications Engineering
In wireless telecardiology applications, electrocardiogram (ECG) signals are often represented in compressed format for efficient transmission and storage purposes. Incorporation of compressed ECG based biometric enables faster person identification as it by-passes the full decompression. This study presents a new method to combine ECG biometrics with data compression within a common JPEG2000 framework. To this end, an ECG signal is considered as an image and the JPEG2000 standard is applied for data compression. Features relating to ECG morphology and heartbeat intervals are computed directly from the compressed ECG. Different classification approaches are used for person identification. Experiments on standard ECG databases demonstrate the validity of the proposed system for biometric identification with high accuracies on both healthy and diseased subjects.
CHANNEL-OPTIMIZED ERROR MITIGATION FOR DISTRIBUTED SPEECH RECOGNITION OVER WIRELESS NETWORKS
(2009)
Lee, Cheng-Lung
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Chang, Wen-Whei
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電信工程研究所
;
Institute of Communications Engineering
This paper investigates the error mitigation algorithms for distributed speech recognition over wireless channels. A MAP symbol decoding algorithm which exploits the combined a priori information of source and channel is proposed. This is used in conjunction with a modified BCJR algorithm for decoding convolutional codes based on sectionalized code trellises. Performance is further enhanced by the use of the Gilbert channel model that more closely characterizes the statistical dependencies between channel bit errors. Experiments on Mandarin digit string recognition task indicate that our proposed mitigation scheme achieves high robustness against channel errors.
Compressed Domain ECG Biometric Identification using JPEG2000
(2015-01-01)
Wu, Yi-Ting
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Wu, Hung-Tsai
;
Chang, Wen-Whei
;
電信工程研究所
;
Institute of Communications Engineering
In wireless telecardiology applications, electrocardiogram (ECG) signals are often represented in compressed format for efficient transmission and storage purposes. Incorporation of compressed ECG based biometric enables faster person identification as it by-passes the full decompression. This study presents a new method to combine ECG biometrics with data compression within a common JPEG2000 framework. To this end, ECG signal is considered as an image and the JPEG2000 standard is applied for data compression. Features relating to ECG morphology and heartbeat intervals are computed directly from the compressed ECG. Different classification approaches are used for person identification. Experiments on standard ECG databases demonstrate the validity of the proposed system for biometric identification with high accuracies on both healthy and diseased subjects.
Compressed Domain ECG Biometric with Two-Lead Features
(2016-01-01)
Lee, Wan-Jou
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Chang, Wen-Whei
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電信工程研究所
;
Institute of Communications Engineering
This study presents a new method to combine ECG biometrics with data compression within a common JPEG2000 framework. We target the two-lead ECG configuration that is routinely used in long-term heart monitoring. Incorporation of compressed-domain biometric techniques enables faster person identification as it by-passes the full decompression. Experiments on public ECG databases demonstrate the validity of the proposed method for biometric identification with high accuracies on both healthy and diseased subjects.
Distributed Source Coding for Mobile ECG Signal Monitoring
(2012-01-01)
Wu, Hung-Tsai
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Tsai, Wei-Ying
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Chang, Wen-Whei
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電信工程研究所
;
Institute of Communications Engineering
This paper describes the design and implementation of an ECG signal monitoring system that is compatible with existing mobile telecommunications networks. First, a two-dimensional ECG signal compression scheme is presented using the distributed source coding (DSC). Also proposed is a modified BCJR algorithm which further enhances the DSC decoding by exploiting the symbol-level a priori information. Finally, we implement a novel integrated system comprising a mobile telephone, with sensors for patient monitoring, and a remote receiver linked by cellular networks. The system, when fully developed, will become powerful aids to monitoring a patient for mobile telemedicine and for emergency situations.
Distributed Source Coding for Real-Time ECG Signal Monitoring
(2014-08-01)
Wu, Hung-Tsai
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Tsai, Wei-Ying
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Chang, Wen-Whei
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電信工程研究所
;
Institute of Communications Engineering
Wireless patient monitoring is an active research area with the goal of ubiquitous health care services. This study presents a novel means of exploiting the distributed source coding (DSC) in low-complexity compression of ECG signals. We first convert the ECG data compression to an equivalent channel coding problem and exploit a linear channel code for the DSC construction. Performance is further enhanced by the use of a correlation channel that more precisely characterizes the statistical dependencies of ECG signals. Also proposed is a modified BCJR algorithm which performs symbol decoding of binary convolutional codes to better exploit the source's a priori information. Finally, a complete setup system for online ambulatory ECG monitoring via mobile cellular networks is presented. Experiments on the MIT-BIH arrhythmia database and real-time acquired ECG signals demonstrate that the proposed system outperforms other schemes in terms of encoder complexity and coding efficiency.
Iterative Source-Channel Decoding Using Symbol-Level Extrinsic Information
(2010-12-01)
Wu, Chun-Feng
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Chang, Wen-Whei
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電信工程研究所
;
Institute of Communications Engineering
Transmission of convolutionally encoded source codec parameters over noisy channels can benefit from the turbo principle through iterative source channel decoding We first formulate a recursive implementation based on sectionalized code trellises for MAP symbol decoding of binary convolutional codes Performance is further enhanced by the use of an interpolative softbit source decoder that takes into account the channel outputs within an interleaving block Simulation results indicate that our proposed scheme allows to exchange between its constituent decoders the symbol level extrinsic information and achieves high robustness against channel noises
Iterative Symbol Decoding of Variable-Length Codes with Convolutional Codes
(2016-02-29)
Wu, Hung-Tsai
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Wu, Chun-Feng
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Chang, Wen-Whei
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傳播研究所
;
Institute of Communication Studies
In this paper, we present a symbol-level iterative sourcechannel decoding (ISCD) algorithm for reliable transmission of variable-length codes (VLCs). Firstly, an improved source a posteriori probability (APP) decoding approach is proposed for packetized variable-length encoded Markov sources. Also proposed is a recursive implementation based on a three-dimensional joint trellis for symbol decoding of binary convolutional codes. APP channel decoding on this joint trellis is realized by modification of the Bahl-Cocke-Jelinek-Raviv algorithm and adaptation to the non-stationary VLC trellis. Simulation results indicate that the proposed ISCD scheme allows to exchange between its constituent decoders the symbol-level extrinsic information and achieves high robustness against channel noises.
JOINT PLAYOUT AND FEC CONTROL FOR ENHANCING PERCEIVED QUALITY OF MULTI-STREAM VOICE COMMUNICATION
(2011)
Chang, Yung-Le
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Wu, Chun-Feng
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Chang, Wen-Whei
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傳播研究所
;
Institute of Communication Studies
A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a joint playout buffer and FEC adjustment scheme that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that the proposed scheme achieves significant reductions in delay and packet loss as well as improved speech quality.
Joint playout and FEC control for multi-stream voice over IP networks
(2013)
Wu, Chun-Feng
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Chiang, Yuan-Chuan
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Chang, Wen-Whei
;
傳播研究所
;
Institute of Communication Studies
Packet loss and delay are the major network impairments for transporting real-time voice over internet protocol (IP) networks. In the proposed system, multiple descriptions of the speech are used to take advantage of packet path diversity. A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a joint playout buffer and forward error control (FEC) adjustment scheme that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that the proposed multi-stream voice transmission scheme achieves significant reductions in delay-and packet-loss rates as well as improved speech quality.
Perceptual-Based Playout Mechanisms for Multi-Stream Voice over IP Networks
(2011-05-01)
Wu, Chun-Feng
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Chang, Wen-Whei
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Chiang, Yuan-Chuan
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電信工程研究所
;
Institute of Communications Engineering
Packet loss and delay are the major network impairments for transporting real-time voice over IP networks. In the proposed system, multiple descriptions of the speech are used to take advantage of the packet path diversity. A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a multi-stream playout buffer algorithm, together with an adaptive parameter adjustment scheme, that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that, compared to FEC-protected single-path transmission, the proposed multi-stream transmission scheme achieves significant reductions in delay and packet loss rates as well as improved speech quality.
Perceptual-Based Playout Mechanisms for Multi-Stream Voice over IP Networks
(2007)
Wu, Chun-Feng
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Lee, Cheng-Lung
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Chang, Wen-Whei
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電信工程研究所
;
Institute of Communications Engineering
cPacket loss and delay are two essential problems to real-time voice transmission over best-effort packet networks. In the proposed system, multiple descriptions of the speech are transmitted to take advantage of largely uncorrelated delay and loss characteristics on different network paths. Adaptive playout scheduling of multiple voice streams is formulated as an optimization problem leading to a better delay-loss tradeoff. Also proposed is a perceptually motivated optimization criterion based on a simplified version of the ITU-T E-model. Experimental results show that the proposed playout buffer algorithm improves the delay-loss tradeoff as well as speech reconstruction quality.
Symbol-Based Joint Decoding of Convolutionally Encoded Variable-Length Codes
(2012-01-01)
Wu, Hung-Tsai
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Lin, Chih-Hsien
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Chang, Wen-Whei
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電信工程研究所
;
Institute of Communications Engineering
In this paper, we present a symbol-level joint source-channel decoding (JSCD) algorithm for transmission of convolutionally encoded variable-length codes (VLCs). The proposed algorithm is based on a two-dimensional state representation which leads to a three-dimensional code trellis. JSCD on this trellis is realized by employing a modified BCJR algorithm which allows to fully exploit the symbol-level source a priori information. Experimental results indicate that the proposed JSCD scheme achieves high robustness against channel noises.
The Role of Accent and Grouping Structures in Estimating Musical Meter
(2020-04-01)
Lin, Han-Ying
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Huang, Chien-Chieh
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Chang, Wen-Whei
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Chien, Jen-Tzung
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電機工程學系
This study presents a new method to exploit both accent and grouping structures of music in meter estimation. The system starts by extracting autocorrelation-based features that characterize accent periodicities. Based on the local boundary detection model, we construct grouping features that serve as additional cues for inferring meter. After the feature extraction, a multi-layer cascaded classifier based on neural network is incorporated to derive the most likely meter of input melody. Experiments on 7351 folk melodies in MIDI files indicate that the proposed system achieves an accuracy of 95.76% for classification into nine categories of meters.