Instruction Set Extensions for MPEG-4 Video

This paper describes instruction set extensions for the acceleration of MPEG-4 algorithms on programmable (RISC-) CPUs. MPEG-4 standardizes audio and video compression schemes for a variety of bit rates and scenarios. As MPEG-4 targets a much broader range of different applications than previously defined hybrid video coding standards like H.263 or MPEG-2, it employs a much higher number of different algorithms and coding modes. Therefore, MPEG-4 implementations will require a more software-oriented approach to be efficient. However, the total computational load for an optimized implementation of an MPEG-4 video codec is expected to exceed the performance levels of today's multimedia signal processors, making further hardware acceleration a necessity. For that purpose, we propose a number of instruction set extensions that add function-specific blocks to the data path of a CPU. These dedicated modules are highly adapted to the most computation-intensive processing schemes of MPEG-4, such as DCT, motion compensation, padding, shape coding, or bitstream parsing. The increased functionality of basic instructions results in a significant speed-up over standard RISC instruction sets, thus making MPEG-4 implementations feasible on programmable processor platforms. Possible target architectures include VLIW multimedia processors, MIMD-style multiprocessors, or coprocessor architectures     

MPEG digital audio coding

The Moving Pictures Expert Group (MPEG) within the International Organization of Standardization (ISO) has developed a series of audio-visual standards known as MFEG-1 and MPEG-2. These audio-coding standards are the first international standards in the field of high-quality digital audio compression. MPEG-1 covers coding of stereophonic audio signals at high sampling rates aiming at transparent quality, whereas MPEG-2 also offers stereophonic audio coding at lower sampling rates. In addition, MPEG-2 introduces multichannel coding with and without backwards compatibility to MPEG-1 to provide an improved acoustical image for audio-only applications and for enhanced television and video-conferencing systems. MPEG-2 audio coding without backwards compatibility, called IMPEG-2 Advanced Audio Coding (AAC), offers the highest compression rates. Typical application areas for MPEG-based digital audio are in the fields of audio production, program distribution and exchange, digital sound broadcasting, digital storage, and various multimedia applications. We describe in some detail the key technologies and main features of MPEG-1 and MPEG-2 audio coders. We also present the MPEG-4 standard and discuss some of the typical applications for MPEG audio compression     

High-quality coding of telephone speech and wideband audio

Digital speech technology is reviewed, with the emphasis on applications demanding high-quality reproduction of the speech signal. Examples of such applications are network telephony, ISDN terminals for audio teleconferencing, and systems for the storage of audio signals, which include the important subclass of wideband speech. Depending on the application, the bandwidth of input speech can vary from about 3 kHz to nearly 20 kHz. Coding for digital telephony at 4 and 8 kb/s, network quality coding at 16 kb/s, and coding for audio at 7 and 20 kHz are examined. Future directions in the field are discussed with respect to anticipated technology applications and the algorithms needed to support these technologies     

MPEG-4音频压缩技术的新进展--ALS

在语音信号处理的许多应用中,采用了无损音频压缩的方法.回顾了MPEG-4音频技术的现状,MPEG-4 ALS(Audio Lossless Coding)标准化过程以及编解码技术.详细介绍了编码器中的线性预测、量化、熵编码等模块,展望了MPEG-4 ALS的应用前景.     

无损音频编码标准MPEG-4 ALS

介绍了MPEG-4的音频无损压缩标准草案——MPEG-4 ALS,并且与目前常见的一些无损音频编码算法作比较。     

频带复制技术的分析和测试

研究高质量的高频重建技术是音频压缩编码技术的必然需求.频带复制技术能够大幅提高感知音频编码器的压缩效率,业已经成为MPEG-4 version 3音频扩展标准.频带复制技术只需提取少量参数,就可在接收端重建高频信号,从而在低比特率下能得到更高的音频质量.本文论述了频带复制技术的原理和特点,并对SBR性能进行评测.     

MPEG的声音编码

本文对ＭＰＥＧ的声音编码进行了讲解，首先对ＭＰＥＧ－１的基本算法进行介绍，然后就ＭＰＥＧ－２算法的不同进行了解说。     

MPEG—4 Audio Version2新概念

介绍了ＭＰＥＧ－４音频第２版的新概念,包括容错健壮性,低延迟音频编码,精细的频段分级,参数音频编码,ＣＥＬＰ静音压缩,扩展的ＨＶＸＣ等。通过与第一版的比较,提出了若干改进之处。     

ATSC Video and Audio Coding

In recent decades, digital video and audio coding technologies have helped revolutionize the ways we create, deliver, and consume audiovisual content. This is exemplified by digital television (DTV), which is emerging as a captivating new program and data broadcasting service. This paper provides an overview of the video and audio coding subsystems of the Advanced Television Systems Committee (ATSC) DTV standard. We first review the motivation for data compression in digital broadcasting. The MPEG-2 video and AC-3 audio compression algorithms are described, with emphasis on basic concepts, system features, and coding performance. Next-generation video and audio codecs currently under consideration for advanced services are also presented.     

Tests on MPEG-4 audio codec proposals

During December 1995, subjective tests were carried out by members of the Moving Picture Experts Group (MPEG, ISO/JTC1/SC29/WG11) to select the proposed technology for inclusion in the audio part of the new MPEG-4 standard. The new standard addresses coding for more than just the functionality of data rate compression. Material coded at very low bit-rates is also included. Thus, different testing methodologies were applied, according to ITU-R Rec. BS 1116 for a bit-rate of 64 kbit/s per channel and according to ITU-T Rec. P.80 for lower bit-rates or functionalities other than data rate compression. Proposals were subjectively tested for coding efficiency, error resilience, scalability and speed change: a subset of the MPEG-4 ‘functionalities’. This paper describes how two different evaluation methods were used and adjusted to fit the different testing requirements. This first major effort to test coding schemes at low bit-rates proved successful. Based on the test results, decisions for MPEG-4 technology were made.

This was the first opportunity for MPEG members to carry out tests on the submitted functionalities. In the process, much was learnt. As a result, some suggestions are made to improve the way new functionalities can be subjectively evaluated.     

Codage audio haute qualité (20 Hz-15 kHz) en sous bandes à débit réduit (64 kbit/s) et à faible retard (5 ms)

In this paper, we present a new method for high quality audio coding at low delay and low bit rate for telecommunications applications such as audioconfe-rence or videoconference. The developped coder is adapted to code generic audio signals at a bit rate of 64 kbit/s with a delay close to 5 ms in the 20-15000 Hz bandwidth. The method is based on speech coding as well as audio coding concepts. The coder combines subband decomposition of the input signal and LD-CELP techniques. We introduce in this structure of coding a psychoacoustic model which allows to allocate an optimal bit rate on each subband according to perceptual properties of the human hearing. In order to satisfy the bit rate requirement of the psychoacoustic model and to reduce the complexity of such a coding algorithm, we suggested a new method of vector quantization based on lattice quantization. This method allows to quantify the residual signal in the LD-CELP coder and avoid the complexity of the full search. Objective and subjective tests have been made on a test set of audio signals which is a critical sub-set used by ISO. Formal tests showed that the quality of the proposed coder is comparable to the best implementation of the MPEG-1, Layer II, but our solution has the advantage of reaching a very low delay (5 ms).     

MPEG-4 high-efficiency AAC coding

The name MPEG-4 high-efficiency AAC (HE-AAC) refers to a family of recent audio coders that was developed by the International Organization for Standardization/International Electrotechnical Commission (ISO/IEC) Moving Picture Experts Group (MPEG) by subsequent extension of the established Advanced Audio Coding (AAC) architecture. These algorithmic extensions facilitate a significant increase in coding efficiency relative to previous standards and other known systems. Thus, they provide a representation for generic audio/music signals that offers high audio quality also to applications limited in transmission bandwidth or storage capacity, such as digital audio broadcasting and wireless music access for cellular phones. This article presents a compact overview of the evolution, technology, and performance of the MPEG-4 HE-AAC coding family.     

低延迟音频编码及应用

通用的感知音频编码技术能在很低码率下提供透明的音质,但编解码算法延迟大,难以满足两地实时通信的要求,语音编码技术能以低延迟提供良好的语音服务,但不适合处理复杂的音频信号。分析了影响感知音频编解码算法延迟的主要因素,并给出了定量的计算方法,着重论述了MPEG-4 AAC-LD的关键技术、性能及其应用。     

MPEG-4编解码算法及其视频性能分析

3G移动终端要求能够提供包含语音、数据和图像的多媒体业务，为此3GPP制定了第一个适合3G系统的基于电路交换的多媒体终端模型——3G-324M规范，该模型支持MPEG-4视频处理，它具有高压缩比和容错编码特性。在研究MPEG-4编解码算法基础上进行了MPEG-4视频编码的性能测试。     

支持MPEG-4面向对象编码的VLSI模块结构设计

MPEG-4中一个新特性是面向对象编码。为了支持这种特性．MPEG-4标准引入了两种新的算法。重复填充和纹理填充.根据这两种算法的特点,设计了MPEG-4重复填充和纹理填充的VLSI结构。文中的方案在Xilinx ISE6．1i集成开发环境下,采用了VHDL进行了描述,并使用电子设计自动化工具进行了模拟和验证。仿真和综合结果表明,本文设计的VLSI处理器,其逻辑功能完全正确,而且可以在很低的时钟频率下满足MPEG-4 Core Profiles ＆ Level2的实时编码要求,可用于MPEG-4的VLSI实现。     

Fast MPEG-4 Motion Estimation: Processor Based and Flexible VLSI Implementations

MPEG-4 is a new multimedia standard combining interactivity, object-based natural and synthetic digital video, audio and computer-graphics. For the implementation of the video part of the MPEG-4 standard a high degree of flexibility is required, where the motion estimation requires the highest part of the computational power. Therefore, in this paper fast algorithms for MPEG-4 motion estimation are evaluated in terms of visual quality and computational power requirements for processor based implementations. Due to the object-based nature of MPEG-4 also new VLSI architectures for MPEG-4 motion estimation are required. Therefore known motion estimation architectures are evaluated on their capability of being modified for MPEG-4 support. Based on this evaluation a new dedicated, but flexible MPEG-4 motion estimation architecture targeted for low-power handheld applications is presented, which resulted to be advantageous to processor based implementations by magnitudes of order.     

AVS无损音频编码标准——熵编码简介

AVS无损音频编码标准,简称AVS-LS,是AVS工作组制定的用于无损压缩音频数据的新标准。AVS-LS采用线性预测和熵编码来对音频数据进行压缩,其中熵编码器用于对线性预测产生的残差信号进行归一化处理及算术编码。介绍了AVS-LS熵编码器的结构及原理,并对AVS-LS的编码效率进行了测试。测试结果表明：AVS-LS的编码效率略低于MPEG-4 ALS,但高于MPEG-4 SLS和FLAC。     

The MPEG-4 Multimedia Coding Standard: Algorithms, Architectures and Applications

The upcoming MPEG-4 standard provides new possibilities for the compression and presentation of multimedia contents. The main characteristics of MPEG-4 are the object-based coding and representation of an audio-visual scene and the ability to code objects of natural or synthetic origin. These features will enhance existing applications with new functionalities and enable standardised solutions for new applications. This paper provides an overview of the three major parts Systems, Visual and Audio of the new MPEG-4 standard, highlights implementation aspects for some envisaged types of MPEG-4 terminals and describes possible future multimedia application scenarios using MPEG-4 functionalities.