Browsing by Author "Wu, Chun-Feng"

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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
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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
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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.
Sensing Phone Use of Motorcycle Drivers
(2014)
Chen, Jyh-Cheng
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Wu, Chun-Feng
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Chung, Wei-Ho
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Ho, Ping-Fan
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資訊工程學系
Due to safety reasons, using mobile phones while driving is prohibited in many countries. Research has also shown that motorcycle riders are 20 times more likely to be killed in a crash than vehicle occupants. Therefore, it is more critical to restrict the use of mobile phones of motorcycle drivers than car drivers. There are some studies that focus on how to distinguish phone use between a driver and other passengers in a car. The techniques used for cars, however, are not always applicable to motorcycles. In this paper, we propose a way to detect phone use of motorcycle drivers. By using two low-cost Bluetooth emitters, mobile phones of the driver and the passenger can measure the signal strengths and decide their locations. We have conducted extensive experiments with various smartphones. The results show that on average we can achieve 9 6 % accuracy.