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In free viewpoint video (FVV) and 3DTV, the depth image-based rendering method has been put forward for rendering virtual view video based on multi-view video plus depth (MVD) format. However, the projection with slightly different perspective turns the covered background regions into hole regions in the rendered video. This paper presents a depth enhanced image summarization generation model for the hole-filling via exploiting the texture fidelity and the geometry consistency between the hole and the remaining nearby regions. The texture fidelity and the geometry consistency are enhanced by drawing texture details and pixel-wise depth information into the energy cost of similarity measure correspondingly. The proposed approach offers significant improvement in terms of 0.2dB PSNR gain, 0.06 SSIM gain and subjective quality enhancement for the hole-filling images in virtual viewpoint video.

With stereo audio stepping into people's daily lives, spatial parametric techn ology comes to its broad prospects for development. Spatial parameteric technology is a method for stereo compression. MDCT is excluded in the current scheme for spatial cues representation, due to its lacking of phase information and energy conservation. In this paper, Construction of new spatial parameter codec framework in MDCT domain, and Optimization of calculating for the new module, to reduce computational complexity of the system.Through the research, implementation, and testing of this two parts, the new spatial parameter codec system, which is constructed in this subject, completes all the basic functions of spatial para metric technology. To compare with the EaacPlus system, the output stereo of the new system gains a considerable subjective and objective test scores. This research makes a contribution to the spatial parametric technology in MDCT domain, and provides a new train of thought to reduce complexity of EaacPlus system.

Audio bandwidth extension (BWE) has emerged as an important tool for the satisfactory performance of low bitrate audio and speech codecs. In the existing BWE method, the high frequency (HF) excitation signals are generated by replicating the low frequency (LF) band directly. However, the coding perception quality will degrade if the correlation between LF and HF bands becoming weak. In this paper, we proposed a new algorithm to restore the HF excitation signals using audio super-resolution. The experiments shown the new algorithm have an outstanding performance for rebuilding HF excitation signals compare with the conventional replication method. In addition, we also provided a new BWE scheme based on audio super-resolution. According to our experimental results, in compare with LPC-based BWE, the subjective listening quality increased by 13% under the same bitrates; in compare with eSBR, the bitrates drop by 63.7% and have the approximate subjective listening quality.

With advanced prediction modes of intra prediction, intra coding of H.264/AVC offers significant coding gains compared with previous video coding standards. It uses an important tool called Lagrangian rate-distortion optimization (RDO) technique to decide the best coding mode for a block, but the computational burden is extremely high. In this paper, we proposed an improved fast intra prediction algorithm including block type selection and mode decision algorithm based on analysis of edge feature of a block. Our algorithm filters out unlikely block type and candidate modes to reduce the RDO calculations. Experimental results showed that the proposed algorithm can reduce the computation complexity of intra prediction from 52.90% to 56.31%, with 0.04 dB PSNR degradation and 2% increase of bit rate.

The error concealment is an important technique in the video transmission,and it can ensure the reconstruction video quality and efficiently recover the data loss and the data errors in the transmission process caused by severe transmission environments.In order to enhance the error resilience of AVS-P2,the paper proposed a new adaptive temporal-spatial error concealment method based on the redundancy motion vectors.To conceal a lost block,the paper used the spatial error concealment for the I-frame macroblocks,and used the temporal error concealment for the non-I-frame macroblocks.At the same time,according to the motion intensity of the macroblocks,it used the default error concealment of AVS-P2 and error concealment method based on redundancy motion vectors,respectively.Lastly,the proposed algorithm was realized based on the platform of the AVS-P2 RM52_20080721.The simulation results show that the proposed method is significantly better than the existing techniques in terms of both objective and subjective quality of reconstruction video.

Mobile audio services are growing with rising popularity of smart mobile devices using WiFi or cellular networks. A major issue facing mobile audio quality assessment is occasional background noises due to the prospect of sound recording at anytime and anywhere with smart mobile devices. Psychological study reveals that people pay selective attention to their interested sound in complex auditory input. In this paper, we model the mobile audio objective quality assessment based on auditory attention mechanism, with attention based horizontal azimuth parameters and timbre distortion parameters as additional Model Output Variables (MOVs). The results show that the prediction accuracy can be obtained by using such a method.

The rapid development of 3D video stimulates demand for 3D audio and promotes the research on perceptual characteristics in 3D sound field. The traditional researches of 3D orientation characteristics were measured sensitivity for some specific locations on site. The test efficiency and results are not good to practical application. This paper first designed a new device to collect optional position in order to establish a 3D sound database and proposed a test method which can be large-scale test. It can ensure the consistency of the experimental data and can't interfere by the experiment personnel. It can improve the efficiency and accuracy of the experiment by using an adaptive method on the computer. It is a significant way to explore the perceptual mechanism of 3D orientation by discovering the relative contribution of frequency and position to selectivity for azimuth and distance. It will provide a theoretical support for the 3D audio acquisition, coding, reconstruction and playback by carrying out the research of 3D orientation characteristics.

With stereo audio stepping into people's daily lives, spatial parametric technology comes to its broad prospects for development. This paper provides the audio bandwidth extension encoding and decoding devices, the bitrate of less high-frequency signal reconstruction, in order to improve the quality of the output audio signal. This research makes a contribution to the lack of existing technology, audio bandwidth extension encoding and decoding devices, digital audio decoding to take advantage of a lower bit rate to restore the original audio signal in the digital encoding process the loss of high-frequency components, thereby enhancing the sound playback quality.

Vector-based amplitude panning in three dimensional sound reproduction aims to preserve both sound image direction and distance perception. While in the estimation process, the loudspeakers are supposed to place on a sphere. It is possible that this requirement cannot be met in home environment. An alternative method to estimate gain factors in vector-based amplitude panning is proposed to preserve distance perception in this study. The experiments confirm that listeners do not perceive obvious distance differences when panning and confirm the validation of the proposed method.

It is important to enhance the resolution of face images from video surveillance for recognization and other post processing. In this paper, a novel sparse representation based face image superresolution (SR) method is proposed to reconstruct a high resolution (HR) face image from a LR observation. First, it gets a HR-LR dictionary pair for certain input LR patch via position patch clustering and locality preserving projection (LPP). LPP is used to further refine the training samples by taking into account the geometry structure between training patches. It can make the dictionary more expressive. Second, it sparse codes the input patch over LR dictionary by solving L1-regularized least squares optimization. F eature-sign algorithm is used to get the stable solution. Third, it maps the sparse coding coefficitons obtain from the LR dictionary to the HR dictionary, and reconstruct the HR face image patch with the coefficitions and HR dictionary. By integrating the hallucinated HR patches together with overlapped in adjacent, we can obtain the final HR face image. Experiments conducted on CAS-PEAL-R1 database validate the proposed method both in subjective and objective quality.

In order to estimate the quality of degraded audio processed by wireless communication networks, a new paradigm has to be designed since conventional objective audio quality assessment methods such as PEAQ are not targeting at the applications. This paper presents a new intrusive objective audio quality assessment method using localized loudness difference based Jitter Distortion Measure (JDM) approach for Quality Assessment of AMR_WB+ (Adaptive Multi-Rate Wideband Plus) codec. We show that including Jitter Distortion Measures and Energy Equalization parameters as additional Model Output Variables (MOVs) of the PEAQ basic version show superior performance in the mobile audio quality assessments.

The features of non-voice signals on public communication nets are analyzed and an approach to distinguish such signals is introduced in this paper. It is the first time for the system to meet three key targets: large-scale, high recognition rate and low false alarm at the same time.

P2P network is a focus research field with rapid development in recent years. Distributed hash tables (DHT for short) schemes are the core technique to locate and search resources in structured P2P network. However, due to the characteristic of DHT that its hash method destroy data locality, DHT only support exact-match search, cannot support complex search process. A simple and flexible solution has been built to tackle the problem which is known as over-DHT paradigm without need to modify existing DHT infrastructures which significantly decreases the complexity of deployment. Among the well known over-DHT paradigms now, LIGHT is an elegant one. It provides more efficient complex query methods than other paradigms and also reduces the paradigm maintenance cost. However, LIGHT does not make full use of the advantages of space partition tree such as that it is a complete binary tree and divides data space into uniform interval in each level of the tree. In this paper, we propose an improved range query algorithm, Upper Boundary Predictably Lookup (UBPL for short) to reduce the DHT-lookup times during range query procedure in LIGHT. We also proved the effectiveness of UBPL by simulating experiments.

The error concealment is an important technique in the video transmission that can assure the reconstruction video quality and can efficiently recovery the data loss and the data errors in the transmission process caused by the transmission environment, thus improve the subjective quality of the video received in the end. In order to enhance the error resilience of AVS-P2 standard, in this paper, we propose a novel adaptive temporal-spatial error concealment method based on the redundancy motion vectors. To conceal a lost block, we use the spatial error concealment for the I-frame macroblocks, and use the temporal error concealment for the non-I-frame macroblocks. At the same time, according to the motion intensity of the macroblocks, we use the default error concealment and based on redundancy motion vectors error concealment method, respectively. Lastly, we realize the algorithm on the platform of the AVS-P2 RM52; the simulations results show that the proposed method is significantly better than the existing techniques in terms of both objective and subjective quality of reconstruction video.

The rate control plays a very important role in video streaming applications. The current rate control scheme in the advance video coding standard H.264/AVC still results in a relatively large quality fluctuation for frame-level with higher motion activities. However, the current video coding standards remove the spatial-temporal redundancy based on the signal statistical correlation. Therefore, to reach higher compression efficiency without perceptually degrading of the reconstructed video signal, the properties of the human visual system need to be better exploited. Research effort has been dedicated to modeling the spatial saliency map based on the properties of HVS. The importance of video image varied with the regions, and human visual attention too varied with the importance of region of video image, so using the attention selectivity properties of the human visual system, the paper proposed video coding rate control method based human visual saliency map in H.264/AVC. For each frame, target bits are allocated through the correction of attention region size between the current frame and previous frame, and for each MB, MB's complexity are modified through the saliency and MAD value of MB. At last, the algorithm was implemented in reference software JM12.2. Simulation results showed that, comparing with traditional algorithm, the proposed reduced the coding bit rate, at the same time, and it also improved the subjective quality of reconstructed video in visual attention region.

In the macroblock coding mode selection process, to improve the judging speed of intra macroblock refresh, the codec usually uses the previous frame temporal information and low-level SVC layer image information to prejudge to reduce the range of candidate macroblock coding modes. As the compressed bitstreams are particular sensitive to error, in order to enhance the error resilience of compressed bitstreams, you can select some inter macroblock mode as the intra macroblock refresh in order to enhance the error resilience performance, to prevent the temporal error propagation. This paper firstly optimizes macroblock level rate-distortion model, and then proposes a candidate modes selection method which compares the rate-distortion function value of intra macroblock mode and inter macroblock mode, to reduce the number of candidate patterns of the rate distortion optimization, to reduce the complexity of the algorithm in the macroblock level, meantime to improve the intra macroblock refreshment rate, effectively prevent t propagation. Under the same bit error rate conditions, the computational complexity is reduced by 15%, and decoded image quality is no significant degrade.

With the rapid development of video communications between heterogeneous networks and video surveillance services, scalable video coding (SVC) technology is proposed to solve the fluctuation of network bandwidth and become the current research focus. Because bit error rate in wireless channel is very high, and it is the serious impact for video quality, the error concealment algorithm research of the scalable video has begun to receive extensive attention. The paper studies the spatial error concealment algorithm for SVC; it will establish frame-level rate-distortion optimization (RDO) model based on scalable video coding framework. While coding the each SVC layer, and planning a flexible slice-packaged strategy for each frame image to establish re-synchronization mechanism in order to prevent the spatial error propagation, as the same bit error rate, the decoded video image quality will improve 0.5~1.5dB. Research findings of the paper will satisfy the requirements of wireless video communications and mobile video surveillance, and so on.

Frequency Domain Linear Prediction (FDLP) gives an approximation of the Hilbert envelopes of a signal. FDLP based Codec works with long temporal segments and keeps the information carried by the time-domain envelopes very well. The codec gives good quality of the reconstructed signal, but coding efficiency is not enough. Here Frequency masking is introduced to FDLP based codec to reduce the bit-rate. Frequency masking is a hearing phenomenon that the hearing threshold of a sound will increase if an intense sound exists simultaneously. The psychoacoustics model is used to estimate the hearing threshold and the absolute threshold of hearing (ATH) of the FDLP carrier signals, and bit allocation for frequency sub-bands FDLP carrier signal is calculated according to the threshold and ATH. 5% bit-rate reduction is obtained with the application of the frequency masking. Perceptual Evaluation of Audio Quality (PEAQ) and Multiple Stimuli with Hidden Reference and Anchor (MUSHRA) test are carried out to evaluate the performance.

In recent years, with the popularity of crime prevention system, which plays a positive role in the field of crime prevention. But a number of security facilities are wasted due to no risk optimization of crime prevention system. This paper firstly establishes a risk optimization model on crime prevention system. Secondly, combinatorial optimization technology is used to calculate this model. Finally, a concrete example is given out to verify the model reasonably. This optimization model can not only guide security facilities more rational allocation and scheduling scientifically and effectively but also fully utilizes the protection capability of crime prevention system. It has great theoretical significance to the planning and layout on crime prevention system in the future.

With the rapid development of industrialization and urbanization, crime prevention issues become serious in China each day. Although various security facilities have played a positive role in the field of crime prevention, defects exist in poorly designed systems targeting personnel protection, physical protection, and electronic protection. Therefore, it is a major strategic task to improve risk assessment techniques and promote the objective evaluation of the risk in such area. In this paper, we present the concept of risk entropy based on the Shannon information theory, derive the risk entropy model of security node, and further analyze the expression and calculation of security node risk entropy. On the other hand, we generate the abstract security network simulation model to study the path dependence of the calculation method on security network risk entropy with a broad crime prevention system formed by connecting security node with traffic network. Finally, we complete the risk assessment of crime prevention system according to the security network. The results show that the risk entropy model is in line with the objective reality. We believe this model can not only break through bottlenecks in the technology of objective risk assessment but also bear important scientific significance in the applications of scientific prediction and effective prevention for public safety emergency response system.

Carrier frequency synchronization is the critical technique in orthogonal frequency division multiplexing (OFDM) systems. In this paper, an improved carrier frequency offset (CFO) estimation method is proposed, which divides the Schmidl’s CFO estimation into coarse and fine estimation by using PN sequence iteration, the new method has higher estimation accuracy and larger estimation range than that of Schmidl’s method in presence of the same computational complexity. The theoretical low bound of CFO estimation is derived by two mathematical methods. The validity of the new algorithm is verified in AWGN and multipath fading channels.

In this paper, structural modifications to the current spatial stereo coding are developed, and two novel complete structures are proposed based on those modifications. Through analyzing the problems of frequency band partition, algorithmic delay, and computational complexity of the current spatial stereo coding on a structural level, 3 schemes, analysis-by-synthesis frequency band partition parallel mono and spatial processing, and filter-bank/transform sharing are put forward. They are shown to be able to solve the problems with reasonable restrictions.

Many multicast fingerprint schemes have been proposed to protect the digital content's copyright in multicast environment. In almost all of those schemes, the generation and the embedding of the fingerprint are controlled by the distributors. Since that, the accusation against the charged redistributors, who was the subscriber in some earlier transaction, is objectionable because that the distributor also has access to the fingerprinted copies and, hence, is able to release such a replica on her own. In this paper, we propose an asymmetric multicast fingerprint scheme which is based on the ideal of secret sharing to solve the subscriberpsilas right problem. A trusted third party FCA is employed and the generation of the fingerprint is separated from the embedding of the fingerprint. Our scheme achieves the following properties: (1) Non-frame: the distributor cannot get to know the exact fingerprinted copy that the subscriber receives, Hence the distributor cannot frame the subscriber by forging the copy which the subscriber has got, (2) Non-repudiation: when distributor finds an unauthorized copy, he can identify the guilty subscriber and the subscriber cannot deny, and (3) collusion-secure: the scheme can resist the collusion attack from multi-subscribers. The result is a security, asymmetric multicast fingerprint protocol.

The subjective quality requirement of ROI is not the same as non-ROI There are very many researches on ROI, but they are little research which jointed the overall RDO theory, the rate control, intra macroblock refreshment and ROI. So the paper proposes the RDO intra refreshment algorithm based ROI, and it uses the end-to-end overall distortion theory to analysis and compute the distortion of ROI and non-ROI, and it makes the marcoblock in ROI obtain more intra refreshment chances. As the bit rate is constrained, so the bit rate of the ROI and non-ROI must be reallocated. At last, the algorithm is implemented in reference software JM12.2 of JVT. The simulation results show that the proposed intra refreshment algorithm based ROI obtained a better subjective quality of the reconstructed video than the traditional algorithm in ROI at the decoder.

In this paper, we present a window switching algorithm employed in AVS (audio video coding standard of China) generic audio coding. The algorithm - energy and unpredictability measure based window switching decision (ENUPM-WSD) - achieves high performance to complexity ratio by taking the advantages of both the low complexity of energy based decision in time domain and the high accuracy of unpredictability based decision in frequency domain. It significantly improves encoded audio quality, especially those with large transient portions, while preserves low average computational complexity in AVS generic audio encoder. Due to the merits, it is accepted as a recommendation module of the AVS audio standard.

Video transport over wireless channel poses a great challenge due to the high variability of the network bandwidth. In this paper, to adjust rate in time according to the change of the quality of the wireless channel, we propose a scheme which judge the quality of the wireless channel proactively. The value of signal strength is used to be an important factor for judging the quality of wireless channel. Experiment results show that the averages of BER and PER can be reduced by employing this scheme.

Audio video coding standard (AVS) is the latest audio and video coding standard of China. AVS Part 7 (also known as AVS-M) audio coding tools target mobility applications where perceptual weighting is of great importance for subjective performance. This paper first briefly introduces the general concept of perceptual weighting. Then a candidate perceptual weighting scheme is proposed and testified on AVS-M audio coder. Further optimization on the candidate is illustrated thereafter. Subjective results of the optimized scheme provide hybrid audio quality comparable to the 3 GPP AMR-WB+.

Detecting overlapping genes from multiple gene lists is a frequently used procedure in microarray- based studies. However in previous studies, the intersection operations of gene lists, formally known as the Intersection-Union Tests (IUTs), were performed without being aware of the incurred changes in Type 1 error rate. We presented an IUT adjustment procedure, called Relaxed IUT (RIUT), which is proved to be less conservative and more powerful for combining independent tests. We demonstrated the superiority of RIUT for detecting overlapping genes by simulation and two real toxicogenomic gene expression data sets. Notably, the better power of RIUT enables it to identify overlapping gene sets leading to known related pathways where traditional TUT fails to find.

The ability to adjust the allocated bandwidth of ongoing flows to cope with resource fluctuation of wireless network is becoming increasingly important. In this paper, we describe a price-based bandwidth adaptation scheme for quality-of-service (QoS) provisioning in ad-hoc network. Depending on the network load, the allocated bandwidth of flows is adjusted dynamically such that the QoS of the wireless network can be optimized. The performance of simulation indicates that the proposed scheme can achieve the goal of optimal channel capacity sharing among multi-hop flows fairly.

This paper presents a new algorithm on packet sequence number reordering, lost motion vector restoration, and video frame bitstream boundary determination. Based on these techniques, a MPEG4 decoder with packet loss concealment functionality has been realized. Experimental results show that this scheme can improve the visual quality obviously for MPEG4 compressed video over wireless Internet network's lack of lower layer QoS guarantee

The part III of Audio Video Standard of China (AVS) is the first Hi-Fi audio coding standard in China public domain. Through nonlinear quantization and context-dependent bitplane entropy coding (CBC), it offers distinct features of nonlinear quantization noise adaptation and high efficiency fine grain scalability (FGS), but also imposes great challenges on fixed-point implementation with high precision requirement and stringent computational resource restriction. This paper presents tuned optimization strategies for fixed-point arithmetic, buffer manipulation, entropy decoding, and CBC decoding loop reduction to address the challenges. The demonstration real-time implementation on fixed-point DSP TMS320C5509A shows that our design optimization cuts down computational complexity from over 200 MIPs to 60 MIPs while preserving over 15 effective bits precision for 16-bit PCM output

Today, much multicast fingerprint schemes have been proposed to protect the digital content's copyright in multicast environment. All of those schemes only consider the distributor's right. The generation and the embedding of the fingerprint are controlled by the distributors. Since that, the accusation against the charged redistributor, who was the subscriber in some earlier transaction, is rejected because that the distributor can also get the distributed fingerprinted copy to this subscriber. This is known as the subscriber's right problem. In Emmanuel and Kankanhalli 2003, they propose a scheme to protect both the distributor's copyright and subscriber's rights for digital video multicast. But in their scheme, they never consider the unbinding problem and the privacy of the subscriber. In this paper, we proposed an anonymous multicast fingerprinting scheme, to fix the Emmanuel and Kankanhalli's scheme by solving the unbinding problem, protecting the subscriber's privacy through a trusted third party. The result is a security, fair and anonymous multicast fingerprint protocol

In this paper, we proposed an adaptive fast inter mode decision algorithm to reduce the computation complexity of H.264/AVC encoder. It makes use of spatio-temporal correlation to filter out unlikely modes. Instead of performing all the seven inter prediction modes, only part of them are chosen for motion estimation and rate distortion optimization (RDO) procedure. Adopting a dynamic adjusted threshold makes the proposed method work well in different bit-rate conditions, especially in low bit-rate. Experimental results indicate that the proposed algorithm can achieve 23% time saving in inter mode decision on average while maintaining the similar rate distortion performance

Recently, there have been many advances in the field of wideband (7 kHz) speech coding because of evolutions and new applications in high-quality communication systems. The paper proposes WBSD (wideband bark spectral distance), a wideband speech codec quality measure based on a bark spectral distance objective quality assessment method for narrow-band telephone speech. The frequency band expansion method and accurate loudness calculation are discussed in detail. The variable bit-rate codecs and packet loss are used to simulate the speech quality degradation on communication systems. Correlation analysis between WBSD and DMOS are examined in order to evaluate objective speech quality measures. The experimental results show that the correlation degree reached is more than 0.8 in informal tests.

AVS Audio Coding Standard is the first standard for Hi-Fi audio in China. The framework of AVS Audio was introduced. Many key technologies are described in details, including long/short window switch decision based on energy and unpredictability, integer MDCT for lossless time-frequency transform, square polar stereo coding, and context-dependent bitplane coding for scalable entropy coding. The informal subject test result is given between AVS audio codec and several dominating audio codecs. It is shown that AVS audio codec is enough for Hi-Fi audio applications.