卫星数字声广播信源编码CMP3和AAC的比较

最近10余年基于感知音频编码的高质量Hi-Fi音频压缩方法已取得突破，本文简要介绍感知音频编码原理，然后分别叙述MP3和MPEG－2 AAC的算法，并对高质量音频编码的主观评价方法作了介绍，给出了MP3和AAC的主观评价结果，在128kbps/2-channel时AAC的重建音质明显高于MP3，最后给出了AAC的其它特点。     

高频重建技术SBR的研究与实现

AAC是MPEG-4标准的音频编码规范，为了提高音频数据的压缩率，往往忽略了音频中所包含的高频成分，只对低频部分进行编码，因此AAC解码器设计中高频重建技术对提高音频质量起着非常重要的作用．本文论述了高频重建技术的原理，并采用SBR技术，实现了AAC音频编码的高频重建，同时性能评测结果表明所实现的AAC SBR系统满足实际应用的要求．     

低码率高性能自然音频编码技术

MPEG-4是近来受到关注最多的视音频压缩编码标准。其自然音频编码部分的MPEG-4 AAC高级音频编码方案采用若干策略取得了比MP3更高的压缩比和更优越的压缩性能。本文详细描述了MPEG-4AAC独特的编码技术,并与MP3的编码性能进行了比较。     

广播与广播系统

0310099卫星数字声广播信源编码 MP3和 AAC 的比较[刊]/陈健//广播与电视技术.—2002,29(12).—135～139(L)最近10余年基于感知音频编码的高质量 Hi-Fi音频压缩方法已取得突破,本文简要介绍感知音频编码原理,然后分别叙述 MP3和 MPEG-2 AAC 的算法,并对高质量音频编码的主观评价方法作了介绍,给出了 MP3和 AAC 的主观评价结果,在128kbps/2-chan-nel 时 AAC 的重建音质明显高于 MP3,最后给出了     

AAC:21世纪音频编码的主流

数字音频编码技术无论对多媒体通信、多媒体广播、消费类电子都有其重要性。首先简要地回顾了数字音频编码技术的发展过程,然后介绍了包括MPEG音频,MP3,AC-3,ATRAC,PAC在内的8种现行商用数字音频编码系统,并作了特征比较,作为重点、详细地叙述了MPEG-2AAC的算法和特点。介绍了ITU-R有关高质量音频编码主观评价的方法,给出了AAC,PAC,MP3,AC-3音质评分表,评分显示在相同低数码率时AAC具有最好的音质,达到了ITU-R的有关规定。最后简要介绍了AAC的实现。     

卫星数字声广播信源编码MP3和AAC的比较

最近 10余年基于感知音频编码的高质量 Hi Fi音频压缩方法已取得突破 ,本文简要介绍感知音频编码原理 ,然后分别叙述 MP3和 MPEG 2 AAC的算法 ,并对高质量音频编码的主观评价方法作了介绍 ,给出了 MP3和 AAC的主观评价结果 ,在 12 8kbps/ 2 channel时AAC的重建音质明显高于 MP3,最后给出了 AAC的其它特点。     

因特网音频编码技术

文章叙述了因特网音频编码技术。首先介绍了感知音频编码原理 ,然后介绍了当前的商用音频编码系统 ,包括MPEG、AC 3、PAC等。最后简要介绍了InternetAudio所采用的标准和技术 ,包括WMA、ePAC、RealAudio、QuikTime、MP3、AAC等。     

MPEG-2/4 AAC 音频编码器实时性能优化研究进展

目前,MPEG-2/4 AAC已成为广泛使用的一种音频编码压缩标准,但是MPEG-2/4 AAC标准的音频编码器存在算法复杂度高、计算量和时闻延迟大等缺点,无法满足实时性要求.从MPEG-2/4 AAC音频编码器的内部结构入手,分析了制约其实时性能的主要环节,就心理声学模型、滤波器组、量化与编码模块3个部分,归纳了当前MPEG-2/4 AAC实时音频编码器实现与优化研究的相关内容和进展,给出了若干算法的主要思想、改进结果和优缺点.在此基础上,提出了进一步优化MPEG-2/4 AAC实时性能的研究方向,以及需要解决的主要问题.     

UMTS音频编码标准研究

未来的编码技术需要将信号处理方面的进展与局部频谱信息技术结合起来。降低计算复杂性是编解码算法演进的主要推动力。文章给出了音频编解码的目标,详细分析了音频编码的原理,研究了MPEG-1、MPEG-2 BC、MPEG-2 AAC、MPEG-4低时延AAC、MPEG-4高效AAC和增强的MPEG-4高效AAC的应用场合、抽样频率和比特率。     

可分级无损音频编码技术的新发展

AAZ（Advanced Audio Zip）是一种基于整数变换的无损音频压缩技术，它能提供从有损到无损的可分级编码，且能与MPEG AAC系统兼容。介绍了AAZ的整体技术框架，并对其关键技术——整数改进离散余弦变换、残差映射和位平面编码作了深入分析，最后与其它无损音频压缩技术在性能上进行了比较，结果表明其具有良好的压缩性能。     

Perceptual Audio Coding Using Sinusoidal/Optimum Wavelet Representation

A perceptual audio coder, in which each audio segment is adaptively analyzed using either a sinusoidal or an optimum wavelet basis according to the time-varying characteristics of the audio signals, has been constructed. The basis optimization is achieved by a novel switched filter bank scheme, which switches between a uniform filter bank structure (discrete cosine transform) and a non-uniform filter bank structure (discrete wavelet transform). A major artifact of the International ISO/Moving Pictures Experts Group (MPEG) audio coding standard (MPEG-I layers 1 and 2) known as pre-echo distortion which uses a uniform filter bank structure for audio signal analysis, is almost eliminated in the proposed coder. A perceptual masking model implemented using a high-resolution wavelet packet filter bank with 27 subbands, closely mimicking the critical bands of the human auditory system, is employed in this audio coder. The resulting scheme is a variable bit-rate audio coder, which provides compression ratios comparable to MPEG-I layers 1 and 2 with almost transparent quality.     

Low bit-rate audio coder using an M-band multiresolution filterbank

A new audio coding system is proposed. Using an M-band multiresolution filter bank technique. This consists of a cascade of 4-band and 8-band filter banks. Experiments with a complete audio coding system were carried out with the proposed filter bank, masking model, bit allocation algorithm, scalar quantisation and Huffman coding. For the broadband signals tested, the proposed system resulted in near transparent quality at bit-rates of 78-91 kbit/s with low computational load. It also achieved similar performance to the MPEG layer 2 coder at 128 kbit/s     

一种结合正弦参数提取和MDCT滤波器组的频谱优化方法

通过研究信号中正弦成分对改进的离散余弦变换(MDCT)的频谱产生的影响，针对使用MDCT滤波器组的音频编码系统，结合正弦参数提取，提出了一种对MDCT频谱进行优化的方法。这种方法利用MDCT和离散傅里里变换(DFT)之间的关系，在提取正弦参数的基础之上，通过简单线性变换求得MDCT谱，并在频域分离信号中的部分准静态正弦．从而达到优化MDCT频谱的目的。     

3D Color Set Partitioning in Hierarchical Trees

This paper introduces the 3D color set partitioning in hierarchical trees (3D-CSPIHT) low bit rate embedded video coding scheme. The codec exploits the correlation between temporal and spatial wavelet coefficients and the interdependency between luminance and chrominance components to code color video sequences without the need for explicit bit allocation. Besides offering rate scalability, the new codec also produces multi-resolution scalable code streams. The hierarchical variable size block matching motion estimation technique is also integrated to demonstrate the motion estimation option with 3D-CSPIHT. The coding results show that 3D-CSPIHT produces better performance and visual quality compared to 3D-SPIHT.     

从有损到无损的音频编解码框架

无损音频编码技术是用于数字音频数据存档,高质量音频编码的一种重要技术,提出了一种兼容有损编解码器的无损音频编解码系统设计。包括通过有损编码残差进行无损扩展和独立的无损编解码两部分。系统采用了声道去相关、整型提升小波、线性预测、残差处理和算术熵编码等技术。在相当复杂度条件下,达到与国际主流无损音频编码技术相当的压缩性能。     

New Low Bit-Rate Audio Coding Approach Based on Adaptive Wavelet Analysis

This paper describes a new audio coding scheme based on adaptive wavelet analysis that provides transparent audio coding for CD-audio signals at low bit rates (≈1.4 bits/sample per channel). A new perceptual cost function is defined to obtain the best wavelet-packet base for each audio frame. The sharp variations in quantization noise that appear at the border of the frames are minimized by a novel approach that avoids overlapping. The proposed coder guarantees high perceptual quality using filters that generate wavelets of any compact support, because a bit-allocation algorithm that takes into account the equivalent filter frequency responses of the synthesis filter bank branches is used.     

基于MPEG-AAC编码器的压缩域音频增强方法

本文基于MPEG-AAC音频编解码器,提出了一种压缩域的音频增强方法.首先,对含噪音频信号的比特流进行解码,得到含噪音频信号的MDCT系数;然后,利用修正的加权递归平均（Modified Weighted Recursive Averaging,MWRA）方法估计噪声功率;再者,利用基于听觉掩蔽原理的自适应β-阶双曲余弦（COSH）统计模型,对含噪音频的MDCT系数进行增强处理;最后,将增强后的MDCT系数重新量化编码,得到用于解码的增强比特流实验结果表明,本文提出的方法能有效去除AAC解码音频信号中的多种背景噪声,其性能明显优于参考方法.     

AMR-WB+技术性能分析和测试

第三代移动通信系统提供的语音短信、流媒体和数字广播服务有着广阔的商业空间和发展潜力,所以一种高效低码率的音频编码方法对移动通信就有着极其重要的意义.3GPP已经把AHR-WB 格式作为其3G流媒体音频部分的编码标准.在低编码速率范围内(JA10Kbps到24Kbps),AHR-WB 编解码格式对音频的处理表现出独特的性能.本文对AMR-WB 进行较深入的研究,并分析和测试AHR-WB 算法的改进带来的音质改善.     

High-quality audio compression using an adaptive wavelet packetdecomposition and psychoacoustic modeling

This paper presents a technique to incorporate psychoacoustic models into an adaptive wavelet packet scheme to achieve perceptually transparent compression of high-quality (34.1 kHz) audio signals at about 45 kb/s. The filter bank structure adapts according to psychoacoustic criteria and according to the computational complexity that is available at the decoder. This permits software implementations that can perform according to the computational power available in order to achieve real time coding/decoding. The bit allocation scheme is an adapted zero-tree algorithm that also takes input from the psychoacoustic model. The measure of performance is a quantity called subband perceptual rate, which the filter bank structure adapts to approach the perceptual entropy (PE) as closely as possible. In addition, this method is also amenable to progressive transmission, that is, it can achieve the best quality of reconstruction possible considering the size of the bit stream available at the encoder. The result is a variable-rate compression scheme for high-quality audio that takes into account the allowed computational complexity, the available bit-budget, and the psychoacoustic criteria for transparent coding. This paper thus provides a novel scheme to marry the results in wavelet packets and perceptual coding to construct an algorithm that is well suited to high-quality audio transfer for Internet and storage applications