MPEG-2 Encoder Research Articles

Fast Image Block Matching for Image Detection and MPEG Encoding Based on Underlying Symmetries

Image block matching is one of the representative methods for image detection and motion compensation in MPEG. Block matching between two images is a problem of finding symmetry between two images by matching macro blocks that are symmetrical to each other in two given images. The greater the PSNR value, the greater the symmetry of the two images. In the given two images, the two macro blocks with the minimum matching error values are regarded as symmetrical to each other. The classical method of calculating the matching error function for every pixel in the entire search area and choosing the smallest of them guarantees global convergence but requires a lot of computation, especially for large intensities. For this reason, many sparse search methods have been developed to reduce the amount of computation. In this paper, we introduce a gradient descent vector optimization algorithm with guaranteed global convergence to the image block matching problem by utilizing the conceptual symmetry of the vector optimization problem, which is a continuous variable, and the image block matching problem, which is a discrete variable. By blurring the image, we transform the matching cost function closer to being unimodal so that the descent-type algorithm works well. As a result, although the proposed method is simple, it can reduce the amount of computation remarkably and has more robustness for the large displacement of image blocks compared to existing sparse search methods.

Improved Recurrence Plots Compression Distance by Learning Parameter for Video Compression Quality.

As the Internet-of-Things is deployed widely, many time-series data are generated everyday. Thus, classifying time-series automatically has become important. Compression-based pattern recognition has attracted attention, because it can analyze various data universally with few model parameters. RPCD (Recurrent Plots Compression Distance) is known as a compression-based time-series classification method. First, RPCD transforms time-series data into an image called "Recurrent Plots (RP)". Then, the distance between two time-series data is determined as the dissimilarity between their RPs. Here, the dissimilarity between two images is computed from the file size, when an MPEG-1 encoder compresses the video, which serializes the two images in order. In this paper, by analyzing the RPCD, we give an important insight that the quality parameter for the MPEG-1 encoding that controls the resolution of compressed videos influences the classification performance very much. We also show that the optimal parameter value depends extremely on the dataset to be classified: Interestingly, the optimal value for one dataset can make the RPCD fall behind a naive random classifier for another dataset. Supported by these insights, we propose an improved version of RPCD named qRPCD, which searches the optimal parameter value by means of cross-validation. Experimentally, qRPCD works superiorly to the original RPCD by about 4% in terms of classification accuracy.

Shading atlas streaming

Streaming high quality rendering for virtual reality applications requires minimizing perceived latency. We introduce Shading Atlas Streaming (SAS), a novel object-space rendering framework suitable for streaming virtual reality content. SAS decouples server-side shading from client-side rendering, allowing the client to perform framerate upsampling and latency compensation autonomously for short periods of time. The shading information created by the server in object space is temporally coherent and can be efficiently compressed using standard MPEG encoding. Our results show that SAS compares favorably to previous methods for remote image-based rendering in terms of image quality and network bandwidth efficiency. SAS allows highly efficient parallel allocation in a virtualized-texture-like memory hierarchy, solving a common efficiency problem of object-space shading. With SAS, untethered virtual reality headsets can benefit from high quality rendering without paying in increased latency.

Mobile edge computing based QoS optimization in medical healthcare applications

Emerging trends in mobile edge computing for developing the efficient healthcare application such as, remote monitoring of the patients with central electronics clouds (e-Clouds) and their increasing voluminous multimedia have caught the attention of everyone in industry and academia. So, clear visualization, big sensing level, and better quality of service (QoS) is the foremost priority. This paper proposes the window-based Rate Control Algorithm (w-RCA) to optimize the medical quality of service (m-QoS) in the mobile edge computing based healthcare by considering the network parameters for instance, peak-to-mean ratio (PMR), standard deviation (Std.dev), delay and jitter during 8 min medical video stream named “Navigation to the Uterine Horn, transection of the horn and re-anastomosis’ transmission over 5 G networks. The performance of the proposed w-RCA is evaluated and compared with the conventional battery smoothing algorithm (BSA) and Baseline by using MPEG-4 encoder for optimizing m-QoS at the source or the server side. The experimental results demonstrate that the w-RCA outperforms the BSA and Baseline by optimizing QoS in remote healthcare application i.e., Tele-surgery. Besides, it is observed and analyzed that w-RCA produces better and effective results at small buffer and window sizes unlike BSA and Baseline by adopting large buffer size during QoS optimization.

Video Quality Assessment through PSNR Estimation for Different Compression Standards

<p align="center"><strong><em>Abstract</em></strong></p><p><em> There is a threefold increase in video traffic over internet. Due to this video compression has become important. Compression of video signals is quiet an interesting task but comes at the cost of video quality. After compression, two methods are scientifically applied to evaluate the quality of video; Subjective and objective analysis. In subjective approach the compressed video is shown to a group of viewers and their feedback is recorded Objective approach aims to set up a mathematical model which can approximate the results of subjective analysis. One such approach is based on the measurement of PSNR. When a signal is applied to the encoder for compression, too much of compression results in a signal with a smaller size but at the same time quality of the signal degrades. In this paper we will compare the quality of compressed video signals produced by H.264, Mpeg2 and Mpeg4 encoder based on the values of MSE and PSNR. Lower the value of MSE, higher will be the PSNR. Comparative plots of MSE, PSNR, SSIM and images for subjective analysis have been added at the end of this paper. </em></p>

Input-Based Dynamic Reconfiguration of Approximate Arithmetic Units for Video Encoding

The field of approximate computing has received significant attention from the research community in the past few years, especially in the context of various signal processing applications. Image and video compression algorithms, such as JPEG, MPEG, and so on, are particularly attractive candidates for approximate computing, since they are tolerant of computing imprecision due to human imperceptibility, which can be exploited to realize highly power-efficient implementations of these algorithms. However, existing approximate architectures typically fix the level of hardware approximation statically and are not adaptive to input data. For example, if a fixed approximate hardware configuration is used for an MPEG encoder (i.e., a fixed level of approximation), the output quality varies greatly for different input videos. This paper addresses this issue by proposing a reconfigurable approximate architecture for MPEG encoders that optimizes power consumption with the goal of maintaining a particular Peak Signal-to-Noise Ratio (PSNR) threshold for any video. Toward this end, we design reconfigurable adder/subtractor blocks (RABs), which have the ability to modulate their degree of approximation, and subsequently integrate these blocks in the motion estimation and discrete cosine transform modules of the MPEG encoder. We propose two heuristics for automatically tuning the approximation degree of the RABs in these two modules during runtime based on the characteristics of each individual video. Experimental results show that our approach of dynamically adjusting the degree of hardware approximation based on the input video respects the given quality bound (PSNR degradation of 1%–10%) across different videos while achieving a power saving up to 38% over a conventional nonapproximated MPEG encoder architecture. Note that although the proposed reconfigurable approximate architecture is presented for the specific case of an MPEG encoder, it can be easily extended to other DSP applications.

A Robust Audio Watermarking Technique Operates in MDCT Domain based on Perceptual Measures

the review presents a digital audio watermarking technique operating in the frequency domain with two variants. This technique uses the Modified Discrete Cosine Transform (MDCT) to move to the frequency domain. To ensure more inaudibility, we exploited the proprieties of the psychoacoustic model 1 (PMH1) of MPEG1 encoder layer I in the first variant and those of psychoacoustic model 2 (PMH2) of MPEG1 encoder Layer III in the second alternative to search the places for insertion of the watermark. In both variants of the technique, the bits of the mark will be duplicated to increase the capacity of insertion then inserted into the least significant bit (LSB). For more reliability in the detection phase, we use an error correction code (Hamming) on the mark. Next, to analyze the performance of the proposed technique, we perform two comparative studies. In the first, we compare the proposed digital audio watermarking technique with her two variants and those achieved by Luigi Rosa and Rolf Brigola, ‘which we download the M-files of each’. The technique developed by Luigi Rosa operates in the frequency domain but using the Discrete Cosine Transform (DCT) as transformation and that proposed by Rolf Brigola uses the Fast Fourier Transform (FFT). We studied the robustness of each technique against different types of attacks such as compression / decompression MP3, stirmark audio attack and we evaluated the inaudibility by using an objective approach by calculating the SNR and the ODG notes given by PEAQ. The robustness of this technique is shown against different types of attacks. In the second, we prove the contribution of the proposed technique by comparing the payload data, imperceptibility and robustness against attack MP3 with others existing techniques in the literature

Run-time parallelization switching for resource optimization on an MPSoC platform

The recent development of multimedia applications on mobile terminals raised the need for flexible and scalable computing platforms that are capable of providing considerable (application specific) computational performance within a low cost and a low energy budget. The MPSoC with multi-disciplinary approach, resolving application mapping, platform architecture and runtime management issues, provides such multiple heterogeneous, flexible processing elements. In MPSoC, the run-time manager takes the design time exploration information as an input and selects an active Pareto point based on quality requirement and available platform resources, where a Pareto point corresponds to a particular parallelization possibility of the target application. To use system's scalability at best and enhance application's flexibility a step further, the resource management and Pareto point selection decisions need to be adjustable at run-time. This research work experiment run-time Pareto point switching for the MPEG-4 encoder. The work involves design time exploration and then embedding of two parallelization possibilities of the MPEG-4 encoder into one single component and enabling run-time switching between these parallelizations, to give run-time control over adjusting performance-cost criteria and allocation deallocation of hardware resources at run-time. The new system has the capability to encode each video frame with different parallelization. The obtained results offer a number of operating points on the Pareto curve in between the previous ones at sequence encoding level. The run-time manager can improve application performance up to 50 % or can save memory bandwidth up to 15 %, according to quality request.

Early-phase performance exploration of embedded systems with ABSOLUT framework

Future interactive embedded systems will support a large number of applications providing users with services related to e.g. telecommunication, audio and video, digital television, internet and navigation. To accommodate these performance demanding applications, the digital processing architectures will evolve from current system-on-chips to massively parallel computers consisting of heterogeneous subsystems connected by a network-on-chip. More flexibility, scalability and modularity are needed from the embedded devices. Consequently, the complexity of system design will increase by orders of magnitude. New methods and tools are needed for the performance evaluation of future embedded systems due to the increasing system complexity. This paper presents a high-level performance modelling and simulation approach called ABSOLUT that alleviates exploration complexity by using abstract virtual system models. The characteristics of the applications are abstracted to workload models that at the bottom level consist of instruction-like primitives. The workload models can be created from application specifications, measurement results, execution traces or source code. The complexity of the execution platform models is reduced since the processing elements need not be modelled in detail and data transfers and storage are simulated only from the performance point of view. The approach enables early evaluation, since the modelling and simulation of complete systems does not require mature hardware or software to exist. ABSOLUT has been applied to a number of case studies including mobile phone usage, MP3 playback, MPEG4 encoding and decoding, 3D gaming, virtual network computing and parallel software defined radio applications. The platforms modelled are either existing or future designs for both embedded systems and personal computers. In several cases, the results obtained from simulations are compared to measurements from real platforms, which reveal an average difference of 12% in the results. This exceeds the accuracy requirements expected from virtual system based simulation approaches intended for early evaluation. In this paper, the most recent enhancements of the ABSOLUT methodology and tool framework are applied in a FFMPEG case study on OMAP4 platform model. The simulation results are compared with those obtained from the execution on an OMAP4-based PandaBoard.

TCTL-preserving translations from timed-arc Petri nets to networks of timed automata

We present a framework for TCTL-preserving translations between time-dependent modeling formalisms. The framework guarantees that once the original and the translated system are in one-by-many correspondence relation (a notion of behavioral equivalence between timed transition systems) then TCTL properties of the original system can be transformed too while preserving the verification answers. We demonstrate the usability of the technique on two reductions from bounded timed-arc Petri nets to networks for timed automata, providing unified proofs of the translations implemented in the verification tool TAPAAL. We evaluate the efficiency of the approach on a number of experiments: alternating bit protocol, Fischer's protocol, Lynch–Shavit protocol, MPEG-2 encoder, engine workshop and medical workflow. The results are encouraging and confirm the practical applicability of the approach.

Estimation of Modelling Parameters for H.263-Quantized Video Traces

We propose methods for selecting the modelling parameters of H.263-quantized video traffic under two different encoding scenarios. For videos encoded with a constant quantization step (unconstrained), we conclude that a two-parameter power relation holds between the exhibited video bit rate and the quantizer value and that the autocorrelation decay rate remains constant for all cases. On the basis of these results, we propose a generic method for estimating the modelling parameters of unconstrained traffic by means of measuring the statistics of the single “raw” video trace. For rate-controlled video (constrained), we propose an approximate method based on the adjustment of the “shape” parameter of the counterpart—with respect to rate—unconstrained video trace. The convergence of the constructed models is assessed viaq-qplots and queuing simulations. On the assumption that the popular MPEG-4 encoders like XVID, DIVX usually employ identical H.263 quantization and rate control schemes, it is expected that the results of this paper also hold for the MPEG-4 part 2 family.

Design and implementation of reconfigurable FIFOs for Voltage/Frequency Island-based Networks-on-Chip

One of the major design bottlenecks in today’s high-performance VLSI systems is the distribution of a single global clock across a chip due to process variability, power dissipation, and multi-cycle cross-chip signaling. A Network-on-Chip architecture partitioned into several Voltage/Frequency Islands (VFIs) is considered as a promising approach for achieving fine-grain system-level power management. In a VFI-based architecture, a clock is utilized for local data synchronization, while inter-island communication is handled asynchronously. To interface the islands on a chip, operating at different frequencies, a complex bi-synchronous FIFO design is inevitable. However, these FIFOs are not needed if adjacent switches belong to the same clock domain. In this paper, a Reconfigurable Synchronous/Bi-Synchronous (RSBS) FIFO is proposed which can adapt its operation to either synchronous or bi-synchronous mode. Four different scalable and synthesizable designs are presented. In addition, a technique is suggested to show how the FIFO could be utilized in a VFI-based NoC. Moreover, we present a mesochronous adaptation method and propose Reconfigurable Mesochronous/Bi-Synchronous (RMBS) FIFOs. Our extensive experiments reveal that compared to a non-reconfigurable system architecture, the proposed reconfigurable FIFOs can help to achieve up to 17% savings in the average power consumption of NoC switches and 29% improvement in the total average packet latency in the case of an MPEG-4 encoder application.

A scalable, non-interfering, synthesizable Network-on-Chip monitor – Extended version

Today’s Multi-Processor System-on-Chips incorporate Network-on-Chips to interconnect multiple processors, memories, and accelerators. We present a freely available toolset to monitor and analyze these networks. Internal signals are pre-analyzed on FPGA without interfering the system. Host PC carries out further analysis with post-processing algorithms and an intuitive graphical interface. Traces of end-to-end communication can be approximated from mere link statistics, average error being 10%. In a case study of MPEG-4 encoder ran at 25MHz, we compared link utilizations and stall cycles by any time window from 500 clock cycles to the whole running time. Area overhead for monitoring was 5%.

Optimising lossless stages in a GPU-based MPEG encoder

Modern GPUs excel in parallel computations, so they are an interesting target to perform matrix transformations such as the DCT, a fundamental part of MPEG video coding algorithms. Considering a system to encode synthetic video (e.g., computer-generated frames), this approach becomes even more appealing, since the images to encode are already in the GPU, eliminating the costs of transferring raw video from the CPU to the GPU. However, after a raw frame has been transformed and quantized by the GPU, the resulting coefficients must be reordered, entropy encoded and framed into the resulting MPEG bitstream. These last steps are essentially sequential and their straightforward GPU implementation is inefficient compared to CPU-based implementations. We present different approaches to implement part of these steps in GPU, aiming for a better usage of the memory bus, compensating the suboptimal use of the GPU with the gains in transfer time. We analyze three approaches to perform the zigzag scan and Huffman coding combining GPU and CPU, and two approaches to assemble the results to build the actual output bitstream both in GPU and CPU memory. Our experiments show that optimising the amount of data transferred from GPU to CPU implementing the last sequential compression steps in the GPU, and using a parallel fast scan implementation of the zigzag scanning improve the overall performance of the system. Savings in transfer time outweigh the extra cost incurred in the GPU.

Intermedia Synchronization Protocol for Continuous Media Using MPEG-4 in Mobile Distributed Systems

The preservation of temporal dependencies among a group of processes that exchange continuous media at runtime is a key issue for emerging mobile distributed systems (MDS), such as monitoring of biosignals and interactive multiuser games. Although several works are oriented to satisfy temporal dependencies, most of them are not suitable for MDSs. In general, an MDS is characterized by the absence of global references (e.g. shared memory and wall clock), host mobility, limited processing and storage capabilities in mobile hosts, and limited bandwidth on wireless communication channels. This paper proposes an asymmetric synchronization protocol to be used at runtime in an MDS without using a common reference. One main aspect of our synchronization protocol is that it translates temporal constraints to causal dependencies of the continuous media data as seen by the mobile hosts. We simulate the protocol by considering a cellular network environment and by using MPEG-4 encoders. The simulation results show that our protocol is effective in reducing the synchronization error. In addition, the protocol is efficient in terms of processing and storage costs at the mobile devices, as well as in the overhead attached per message across the wired and wireless channels.

Realization of QoS Control Strategies in the next Generation Network on the Basis of Differentiated Service Model and Priority of Inter-Frame

With the development of networks and wideband technology, Next Generation Network (NGN) evolves into integrated transmission network of the multimedia information which including data, voice, image, the real-time tasks, etc. The traditional IP network provides only "best-effort" service and cannot meet the needs of multi-media applications and the different demands of a variety of users for network transmission quality. On the basis of a detailed analysis to Differentiated service (DiffServ) model and the characteristics of MPEG-4 encoding, the QoS control strategy of video streaming in NGN is proposed. That is a QoS control strategy which uses priority of Inter-Frame combining with DiffServ model in network congestion. Simulation and testing results show that the strategy can improve the transmission quality of the video stream effectively in the case of limited bandwidth.

Methodology for Energy-Flexibility Space Exploration and Mapping of Multimedia Applications to Single-Processor Platform Styles

Embedded multimedia devices are now a common element of our environment, such as mp3 players, handheld devices, and so on. Choosing the right main processing element is a key issue for the success of these devices, and their consumption, performance, retargetability, and development time are some of the elements that need to be analyzed and well-balanced. In this paper, we map the same multimedia application (MPEG-4 main profile) into various target platforms generally used in the embedded area. The design flow of our work starts with a single MPEG-4 encoder description which is later refined and optimized to be implemented on different platforms: an embedded platform formed by a high performance digital signal processor and an embedded processor, an application specific instruction processor, a specific hardware implemented in a field-programmable gate array for accelerating the data-flow part of the system with a soft-core for the control part, and an application specific integrated circuit. The main contribution of this paper is to illustrate a methodology that can be generalized to different data dominant applications. This paper describes a new methodology to obtain near optimal implementation from concept to silicon for all platforms and it can be extended to any hybrid HW/SW multimedia platform. We evaluate the different transformations of each platform to arrive at an optimal implementation. These higher level transformations allow achieving better results than using more precise efforts in mapping the design in the physical level. This methodology can be extended to any data dominant application.

Linking run-time resource management of embedded multi-core platforms with automated design-time exploration

Nowadays, owing to unpredictable changes of the environment and workload variation, optimally running multiple applications in terms of quality, performance and power consumption on embedded multi-core platforms is a huge challenge. A lightweight run-time manager, linked with an automated design-time exploration and incorporated in the host processor of the platform, is required to dynamically and efficiently configure the applications according to the available platform resources (e.g. processing elements, memories, communication bandwidth), for minimising the cost (e.g. power consumption), while satisfying the constraints (e.g. deadlines). This study presents a flow linking a design-time design space explorer, coupled with platform simulators at two abstraction levels, with a fast and lightweight priority-based heuristic integrated in the run-time manager to select near-optimal application configurations. To illustrate its feasibility and the very low complexity of the run-time selection, the proposed flow is used to manage the processors and clock frequencies of a multiple-stream MPEG4 encoder chip dedicated to automotive cognitive safety applications.

18GHz 지하철 무선영상 전송시스템 개발

지하철 운행 중 차지상간 무선영상전송을 위해 18GHz 대역을 이용한 OFDM 방식의 영상 전송 시스템을 개발하였다. 개발된 시스템은 디지털 변조 보드와 RF 보드로 구성하였으며, 지하철 터널구간 내에서 18GHz의 멀티패스 페이딩에 강한 특성을 가진다. 시스템의 검증을 위하여 스펙트럼 분석기를 사용해 전송 시스템의 출력 신호를 분석하였고, 분석결과 구현된 시스템은 350MHz에 대역폭 8MHz, -19dBm의 채널 파워를 가짐을 확인할 수 있었다. 또한, 영상의 송 수신 특성을 시험하기 위하여 SR사의 MPEG-2엔코더를 사용하여 영상을 입력하고 송신한 후 SR사의 수신기를 이용하여 영상 수신 실험을 하였다. 실험결과 송신한 영상이 손실 없이 수신됨을 확인하였다. The video transmission system using OFDM method at 18GHz frequency band was developed for wireless video transmission from subway to monitoring center when subway running. It is composed of digital modulation board and RF board. And it has a robust characteristics of multipath fading in subway tunnels at 18GHz. To verify the system, we used spectrum analyzer for analysing the output of transmission system and get the results that it has 350MHz frequency, 8MHz bandwidth and -19dBm channel power. Also we tested video receiving using MPEG-2 encoder and got good quality images.

Measurement of IPTV traffic from an operative network

AbstractIn this paper, we present measurement results of IP TeleVision (IPTV) multicast traffic collected from real of the FastWeb network. FastWeb is an ISP provider, which is the main broadband telecommunication company in Italy. The network relies on an IP architecture and, besides television channels, delivers to the user services such as data transfer and VoIP over a single broadband connection. The traffic we focus on consists of 83 high‐quality digital TV channels encoded using different MPEG‐2 encoders. The results show that, depending on the encoder and based on the bitrate, flows can be classified as being: 1‐VBR, i.e. with one typical value of the mean bit rate, 2‐VBR, i.e. two typical bitrate values are observed, and VBR. Measurement of the packet loss, jitter and inter‐packet gap show that, independently from the class, packet generation process of the flows can have various degrees of burstiness. Despite the packet level burstiness, average jitter is limited to few milliseconds and no packet loss was ever observed in the backbone. Moreover, the evolution of inter‐packet gap over time remains basically unchanged when observing a flow in different time periods. Our results prove that the FastWeb network is properly carrying IPTV traffic, and that today technology is mature to make TV over IP feasible and of good quality. Copyright © 2009 John Wiley & Sons, Ltd.