Browsing by Author "Liu, CM"

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A Bayes-Marlsov based maneuver estimator for maneuver target tracking
(1996)
Liu, CM
;
Wu, KG
;
交大名義發表
;
資訊工程學系
;
National Chiao Tung University
This paper proposes a new method for estimating a target's maneuver value. The method consists of a Baysian scheme for identifying the possible range of a target's maneuver magnitude and a multiple-hypothesis scheme for estimating its value. The parameters of the multiple-hypothesis scheme are dynamically adjusted with the variation of a target's maneuvering behavior. The dynamic adjustment helps keep a low complexity while dealing with a large range of target maneuverability Simulation results show that the new method, although has a low computational complexity, can achieve the same accuracy as the original multiple-hypothesis method in [3].
A hierarchical decimation lattice based on N-queen with an application for motion estimation
(2003-08-01)
Wang, CN
;
Yang, SW
;
Liu, CM
;
Chiang, TH
;
資訊科學與工程研究所
We present a novel technique, N-queen lattice, to spatially subsample a block of pixels. Although this lattice is pertinent to many applications, we present an application to speed up motion estimation with minimal loss of coding efficiency. The N-queen lattice is constructed to characterize spatial features in all directions. It can be hierarchically organized for motion estimation With variable nonsquare block size. Despite the randomized lattice structure, we demonstrate that it is possible to achieve compact data storage architecture for efficient memory access and simple hardware implementation. Our simulations show that the N-queen lattice is superior to several existing sampling techniques with improvement in speed by about N times and small loss. in peak SNR.
A hierarchical N-queen decimation lattice and hardware architecture for motion estimation
(2004-04-01)
Wang, CN
;
Yang, SW
;
Liu, CM
;
Chiang, TH
;
資訊工程學系
A subsampling structure, an N-Queen lattice, for spatially decimating a block of pixels is presented. Despite its use for many applications, we demonstrate that the N-Queen lattice can be used to speed up motion estimation with nominal loss of coding efficiency. With a simple construction, the N-Queen lattice characterizes the spatial features in the vertical, horizontal, and diagonal directions for both texture and edge areas. Especially in the 4-Queen case, every skipped pixel has the minimal and equal distance of unity to the selected pixel. It can be hierarchically organized for variable nonsquare block-size motion estimation. Despite the randomized lattice, we design compact data storage architecture for efficient memory access and simple hardware implementation. Our simulations show that the N-Queen lattice is superior to several existing sampling techniques with improvement in speed by about N times and small loss in peak SNR (PSNR). The loss in PSNR is negligible for slow-motion video sequences and is less than 0.45 M at worst for high-motion estimation sequences.
A new hardware-efficient architecture for programmable FIR filters
(1996-09-01)
Lee, HR
;
Jen, CW
;
Liu, CM
;
電子工程學系及電子研究所
;
Department of Electronics Engineering and Institute of Electronics
Although much research has been done on efficient high-speed filter architectures, much of this work has focused on filters with fixed coefficients, such as Canonical Signed Digit coefficient filter architectures, multiplierless designs, or memory-based designs. In this paper, we focus on digit-serial, high-speed architectures with programmable coefficients. To achieve high performance goals, we consider both of algorithm level and architecture implementation level of FIR filters. In algorithm level, we reformulate the FIR formulation in bit-level and take the associative property of the addition in both the digit-serial multiplications and filter formulations. In architecture level, we considered issues to implement the reformulated results efficiently. The issues include addition implementation, data flow arrangements, and treatment of sign-extensions. Based on the above considerations, we can obtain a fitter architecture with accumulation-free tap structure and properties of short latency, flexible pipelinability and high speed. Comparing the cost and performance with previous designs, we find that the proposed architecture reduces the hardware cost of a programmable FIR filter to only half that of previous designs without sacrificing performance.
A new hardware-efficient architecture for programmable FIR filters
(1996-09-01)
Lee, HR
;
Jen, CW
;
Liu, CM
;
電子工程學系及電子研究所
;
Department of Electronics Engineering and Institute of Electronics
Although much research has been done on efficient high-speed filter architectures, much of this work has focused on filters with fixed coefficients, such as Canonical Signed Digit coefficient filter architectures, multiplierless designs, or memory-based designs. In this paper, we focus on digit-serial, high-speed architectures with programmable coefficients. To achieve high performance goals, we consider both of algorithm level and architecture implementation level of FIR filters. In algorithm level, we reformulate the FIR formulation in bit-level and take the associative property of the addition in both the digit-serial multiplications and filter formulations. In architecture level, we considered issues to implement the reformulated results efficiently. The issues include addition implementation, data flow arrangements, and treatment of sign-extensions. Based on the above considerations, we can obtain a fitter architecture with accumulation-free tap structure and properties of short latency, flexible pipelinability and high speed. Comparing the cost and performance with previous designs, we find that the proposed architecture reduces the hardware cost of a programmable FIR filter to only half that of previous designs without sacrificing performance.
A new intensity stereo coding scheme for MPEG1 audio encoder - Layers I and II
(1996-08-01)
Liu, CM
;
Liu, JC
;
交大名義發表
;
資訊工程學系
;
National Chiao Tung University
In MPEG I audio compression standard, the intensity stereo coding scheme is a keg, technology to achieve a bit rate lower than 2x128 kbit/s for stereophonic signals while preserving high audio quality. However, the intensity coding scheme has difficulties in signal cancellation and scalefactor consistency. This paper shows that the Karhunen-Loeve (KL) transform can be applied to the intensity coding for solving the two problems. In this paper, we present true methods to implement the Karhunen-Loeve transform in the MPEG 1 layers I and II. Among the two methods, one is fully compatible with the MPEG1 while the other need a slight coding of modification on the MPEG coding process. All the conclusion. two methods are tested to have better audio quality than the technique indicated by the MPEG 1 layers 1 and II.
A new intensity stereo coding scheme for MPEG1 audio encoder - Layers I and II
(1996-01-01)
Liu, JC
;
Liu, CM
;
資訊工程學系
;
Department of Computer Science
A new intensity stereo coding scheme for MPEG1 audio encoder - Layers I and II
(1996-08-01)
Liu, CM
;
Liu, JC
;
資訊工程學系
;
Department of Computer Science
In MPEG I audio compression standard, the intensity stereo coding scheme is a keg, technology to achieve a bit rate lower than 2x128 kbit/s for stereophonic signals while preserving high audio quality. However, the intensity coding scheme has difficulties in signal cancellation and scalefactor consistency. This paper shows that the Karhunen-Loeve (KL) transform can be applied to the intensity coding for solving the two problems. In this paper, we present true methods to implement the Karhunen-Loeve transform in the MPEG 1 layers I and II. Among the two methods, one is fully compatible with the MPEG1 while the other need a slight coding of modification on the MPEG coding process. All the conclusion. two methods are tested to have better audio quality than the technique indicated by the MPEG 1 layers 1 and II.
A new postprocessing method for the block-based DCT coding based on the convex-projection theory
(1998-08-01)
Liu, CM
;
Wang, CN
;
Lin, JY
;
資訊科學與工程研究所
;
Institute of Computer Science and Engineering
The block-based DCT coding has been adopted in image compression standards such as JPEG and MPEGs. This type of coding causes some noticeable artifacts known as the blocking effect. Various post processing methods have been devoted to removing the blocking effect; however, these methods also unavoidably blur the decoded image. Removing the blocking effect while avoiding blurring is the main issue of the postprocessing. This paper presents a new postprocessing method based on convex-projection theory. This method is shown to have keep good tradeoff for the blocking effects and the blurring effects.
A timestamp-sensitive scheduling algorithm for MPEG-II multiplexers in CATV networks
(1998-09-01)
Lin, YD
;
Liu, CM
;
資訊工程學系
;
Department of Computer Science
To achieve smooth display of MPEG-II programs in the residential cable TV networks, we present a timestamp-sensitive scheduling algorithm for MPEG-II multiplexers. The deadline-driven scheduler maintains, for each program stream, a counter and a timestamp to record and determine how many Transport Stream (TS) packets should be transmitted before the current scheduling cycle ends. The decoding timestamp (DTS) of TS packets is used to update the counter in order to prevent deadline violation. This algorithm is compared numerically with the timestamp-insensitive algorithm which runs constant-bit-rate (CBR) scheduling. The trace-driven simulation shows that the deadline violation of our timestamp-sensitive scheduling is much lower than CBR's and well controlled for programs with various degrees of burstiness. We also show that the algorithm can be further improved by adding a scheme to prevent buffer underflow and overflow at multiplexers and set-top-boxes, respectively.
A unified fast algorithm for cosine modulated filter banks in current audio coding standards
(1999-12-01)
Liu, CM
;
Lee, WC
;
資訊工程學系
;
Department of Computer Science
Current audio coding standards, such as MPEG-I layers 1-3, MPEG-II layers 1-4, MPEG-IV, and AC-3, have adopted for compression various forms of the cosine modulated filter banks (CMFBs). It is demonstrated that all modulated cosine transforms can be divided into two modules-the permutation and discrete cosine transform. The derived discrete cosine transforms are either type II, type III, or type IV. A fast computing algorithm is proposed that will compute all three types of transforms uniformly. The new fast algorithm has good features with regard to regularity, complexity, and general applicability.
Adaptive contention window-based cluster head election mechanisms for wireless sensor networks
(2005)
Wang, LC
;
Wang, CW
;
Liu, CM
;
交大名義發表
;
National Chiao Tung University
The clustering architecture is essential in achieving the goal of energy efficiency for a wireless sensor network. In general, a clustering algorithm consists of the cluster head election and the cluster member assignment mechanism. This paper proposes an adaptive contention window (ACW)-based cluster head election mechanism. Unlike other legacy cluster head election mechanisms such as LEACH (Low Energy Adaptive Clustering Hierarchy) protocol, the proposed ACW algorithm can achieve four major goals in cluster head election for wireless sensor networks: 1) high successful probability of cluster head election, 2) appropriate number of cluster heads, 3) uniform distribution of cluster heads, and, 4) equal times to be a cluster head for each sensor, simultaneously.
An IMM algorithm incorporating a maneuver estimation method for tracking maneuvering targets in airborne radar
(1996)
Liu, CM
;
Wu, KG
;
交大名義發表
;
資訊工程學系
;
National Chiao Tung University
This paper proposes a new maneuvering tracking algorithm which consists of a multiple-hypothesis method for estimating a target's maneuver value and the interactive multiple model (IMM) algorithm for estimating a target's state. The new algorithm contains two motion model: one constant-speed model and one acceleration model. The parameters of the acceleration model will be adapted with the target dynamic based on the maneuver estimation results. Simulation results show that during the linear constant acceleration periods the speed estimation errors of the new algorithm can be reduced to about 50% of the conventional IMM algorithm, which uses 25 motion models of fixed model parameters to cover a maneuver range of -10 similar to 10g.
Bit allocation for advanced audio coding using bandwidth-proportional noise-shaping criterion
(2003)
Liu, CM
;
Lee, WC
;
Chien, CT
;
資訊工程學系
;
Department of Computer Science
The bandwidth-proportional noise-shaping criterion has been proposed to be the optimum bit allocation criterion in the sense of minimizing segmental masking-to-noise ratio [1]. A single loop bit allocation has been developed for ISO/IEC MPEG-1layer 3 (MP3) [2] to reduce the complexity over the existing two nested loop method for MP3 encoder. Also, the allocation method also leads to the better compression quality. This paper extends the allocation method to the ISO/IEC MPEG-2/4 Advanced Audio Coding (AAC) [3] [4]. This paper derives the perceptual bandwidth for the scale factor bands, provides the bit consumption trade-off to transmit the bits for the scale factors, and illustrates an efficient sectioning method on the Huffman codebook. All these methods jointly integrate with the bit allocation method to improve the quality and the computing complexity of the widely used AAC reference encoder, FAAC [5].
Bit allocation method for AC-3 encoder
(1998-08-01)
Liu, CM
;
Lee, SW
;
Lee, WC
;
資訊工程學系
;
Department of Computer Science
Dolby AC-3 is an audio coder selected in the United State high definition television (HDTV) standards and widely adopted for the audio codec in DVD films. For audio coder, the complexity of the audio encoders is always higher than that of the decoders. A main module leading to the higher complexity is the bit allocation process. The objective of the bit allocation is to allocate restricted bits to encoded information. Since that AC-3 coder has adopted a floating point representation which the magnitude of each spectral line is represented with an exponent and a mantissa, the bit allocation process not only has to decide the suitable quantization method for the mantissa similar to the process in other coding standards such as MPEG1, but also the exponent strategies and hence the parameters in psychoacoustic models. This coding process has been referred to as a hybrid coding [5] and has led to immense complexity for an encoder. This paper proposes an efficient method for the bit allocation process.
Cross-layer design for determining the optimal number of clusters in a wireless sensor network
(2004)
Wang, LC
;
Wang, CW
;
Liu, CM
;
交大名義發表
;
National Chiao Tung University
The clustering concept can help a wireless sensor network establish hierarchy among numerous sensor nodes. Clustering architecture is essential in achieving the goal of saving power for wireless sensor networks because a cluster representative node can forward data to sink for the covered cluster area. One of key parameters in the clustering design is the number of clusters in a entire wireless sensor network. To this end, we suggest a cross-layer approach to determine an optimal number of clusters in an observation area from the perspectives of physical (PHY) layer, medium access control (MAC) sub-layer, scheduling (SCH) sublayer and network (NET) layer.
Design of the coupling schemes for the AC-3 coder in stereo coding
(1998-08-01)
Liu, CM
;
Lee, WC
;
Juang, SY
;
資訊工程學系
;
Department of Computer Science
When applying the Dolby AC-3 coder for the stereo music compression, the coupling scheme that combines the two channels stereo audio signals in high frequency into one channel is the key technology for the Dolby AC-3 to achieve the bit rates lower than 96x2 kbits/sec while preserving high stereo audio quality. This paper proposes four coupling methods for the AC-3 encoder. These four methods vary with the complexity and performance. These four methods are compared through both subjective and objective tests. These four coupling methods are also combined with the dithering scheme and examined through subjective and objective tests. The result shows that the dithering scheme can effectively ease the coupling artifacts and enhance the audio quality.
Design of vocabulary-independent Mandarin keyword spotters
(2000-07-01)
Liu, CM
;
Chiu, CC
;
Chang, HY
;
資訊工程學系
;
Department of Computer Science
This paper considers the design of a vocabulary-independent keyword spotter for Mandarin speech according to the framework by Huang et al. [1]. This paper considers three varieties of filler model structures for the framework based on subsyllabic grammar of Mandarin speech. On the basis of the three structures, we infer the problems of this framework through three arguments and presents two methods to modify the spotting vehicle.
Fast motion estimation using N-queen pixel decimation
(2001)
Yang, SW
;
Wang, CN
;
Liu, CM
;
Chiang, TH
;
資訊科學與工程研究所
We present a technique to improve the speed of block motion estimation using only a subset of pixels from a block to evaluate the distortion with minimal loss of coding efficiency. To select such a subset we use a special sub-sampling structure, N-queen pattern. The N-queen pattern can characterize the spatial information in the vertical, horizontal and diagonal directions for both texture and edge features. In the 4-queen case, it has a special property that every skipped pixel has the minimal and equal distance of one to the selected pixel. Despite of the randomized pattern, our technique has compact data storage architecture. Our results show that the pixel decimation of N-queen patterns improves the speed by about N times with small loss in PSNR. The loss in PSNR is negligible for slow motion video sequence and has 0.45 dB loss in PSNR at worst for high motion video sequence.
Fast perceptual convolution for room reverberation
(2003)
Lee, WC
;
Liu, CM
;
Yang, CH
;
Guo, JI
;
資訊工程學系
The FIR-based reverberators, which convolve the input sequence with an impulse response modelling the concert hall, have better quality compared to the IIR-based approach. However, the high computational complexity of the FIR-based reverberators limits the applicability to most cost-oriented system. This paper introduces a method that uses perceptual criterion to reduce the complexity of convolution methods for reverberation. Also, an objective measurement criterion is introduced to check the perceptual difference from the reduction. The result has shown that the length of impulse response can be cut off by 60% without affecting the perceptual reverberation quality. The method is well integrated into the existing FFT-based approach is have around 30% speed-up.