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.     

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．     

基于分数阶滤波器的ADPCM预测误差信号处理

ADPCM音频压缩算法在G．721,G．723和G．726等相关数字语音通信协议中得到广泛应用。针对ADPCM中普遍存在的预测误差信号镜频干扰和高频噪声的问题,提出了基于分数阶积分器的误差信号频谱压缩滤波迭代算法和基于分数阶积分器的逆系统信号恢复迭代算法,为进一步提高音频压缩编码品质提供了新的方法,理论分析和仿真结果表明所提出方法的可行性。     

数字广播音频编码中的频带复制技术(SBR)

介绍数字广播音频编码中的频带复制(SBR)技术,分析在音频编码方面使用SBR的必要性,并且简单介绍了SBR编码器和解码器的工作进程.SBR是一种新型的音频编码技术,可以用相对低的比特率达到高质量的音频效果.     

基于相似关联度神经网络的音频频带扩展

宽带音频带宽的限制会降低其主观质量和自然度.本文提出了一种基于相似关联度神经网络的宽带向超宽带音频频带扩展方法.该方法将宽带音频的精细谱重构成多维相空间,并建立相似关联度神经网络来恢复高频成分的精细谱,同时借助高斯混合模型估计高频谱包络,并以G.722.1编码器为平台实现音频信号的带宽扩展.测试结果表明,本文方法扩展性能优于参考方法,其主观质量接近于G.722.1C超宽带编码器.     

用非均匀时频变换提高音频编码器性能

预回声对音频压缩编码器的性能影响很大。人们一般采用窗切换技术来降低其不利影响。窗切换中短窗提高了对暂态信号的时间分辨率,同时降低了对低频稳态信号的频率分辨率。设计了一种非均匀时频变换方法,它在保留了对信号低频平稳分量的频率分辨率的同时提高了对高频暂态分量的时间分辨率。最后用实验验证了方法的可行性和有效性。     

基于回声状态网络的音频频带扩展方法

宽带音频通信系统对传输信号有效带宽的限制会降低重建音频的主观质量和自然程度.本文提出了一种基于回声状态网络的宽带向超宽带音频盲目式频带扩展方法.该方法借助回声状态网络来模拟音频信号高低频频谱参数间的映射关系,并依据网络模型中的时延递归结构连续更新系统状态来近似描述音频特征的时域演变过程,有效地估计了高频成分的频谱包络.同时,结合频谱复制方法得到的高频频谱细节,该方法实现了宽带向超宽带音频的有效扩展.测试结果表明,本文所提方法提升了宽带音频的听觉质量;对于多数测试数据,该方法在静态和动态失真方面获得了优于高斯混合模型扩展方法的扩展性能.     

AC—2广播与存贮的高质量数字音频编码器

本文介绍了美国杜比实验室研制的适用于广播的宽带数字音频变换编码器系统。编码器的比特率每信道192～64kbit/s,它能提供不同的比特率、编码延时及音频带宽三者折衷的方案。还重点介绍了新一代音频编码器,这种编码器在比特率压缩过程中应用了人耳在频域和时域的掩蔽效应,使用了新型滤波器簇,能根据信号的时间特性和频率特性,动态调整频率与时间分辨率。还介绍了编码器系统现在和将来的应用。     

基于频谱减法的语音去噪算法研究

语音增强技术是音频信号处理中的重要部分,频谱减法是目前在语音增强技术中最常用的方法之一。针对传统频谱减法会产生音乐噪声并无法消除音乐噪声的不足之处及高频噪声干扰比较严重的情况下频谱减法效果差的情况,采用了在频谱减法之后进行LMS滤波以降低音乐噪声对语音质量的影响和低通滤波以滤除脉冲干扰。根据仿真结果表明,改进扩展频谱减法能够有效降低音乐噪声和尖锐的高频兹兹声,从而提高信噪比,达到语音增强的目的。     

一种低码率音频压缩编码的窗切换方法

当前主流音频压缩技术都采用窗切换技术来抑制预回声现象。然而编码器切换到短窗虽然提高了对暂态信号的时间分辨率,但同时却降低了低频平稳信号的频率分辨率。对此提出在短窗时采用一种非均匀时频变换方法来取代短窗时频变换,它对信号的低频平稳分量具有很高的频率分辨率,同时对高频暂态分量也有足够的时间分辨率。最后用实验验证了该方法的可行性和有效性。     

低码率音频编码技术综述

介绍了当今主流低码率音频编码标准及其关键技术,包括EAAC 及其SBR技术,PS技术,AMR-WB 及TCX技术,G.729.1及嵌入式分层编码技术.     

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     

基于ADSP-21261的数字效果器的设计

介绍了采用AD公司的SHARC ADSP-21261组成的数字音频信号处理系统及系统的软硬件设计，并对音频信号的混响处理算法进行了论述。经实际应用证明，该系统是一个不错的音频信号采集与处理系统。     

RAVI, a proposed standard for the interchange of audio/visualinteractive applications

RAVI (representation for audio/visual interactive applications), a set of specifications proposed as a standard for the interchange of audio/visual interactive applications (AVIs), is presented. RAVI was designed to meet the functional requirements of an AVI. RAVI consists of an interchange format and a set of functional operators (formulation). Protocols for a RAVI system are defined to allow communication between distributed subsystems. RAVI profiles allow compatible implementation of restricted sets of functionalities. As far as possible, the RAVI specifications use existing standards and are being submitted to ISO and CCITT for discussion. Products are commercially available, and a number of applications has already been developed     

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.     

Digital audio coding for visual communications

Current and future visual communications for applications such as broadcasting videotelephony, video- and audiographic-conferencing, and interactive multimedia services assume a substantial audio component. Even text, graphics, fax, still images, email documents, etc. will gain from voice annotation and audio clips. A wide range of speech, wideband speech, and wideband audio coders is available for such applications. In the context of audiovisual communications, the quality of telephone-bandwidth speech is acceptable for some videotelephony and videoconferencing services. Higher bandwidths (wideband speech) may be necessary to improve the intelligibility and naturalness of speech. High quality audio coding including multichannel audio will be necessary in advanced digital TV and multimedia services. This paper explains basic approaches to speech, wideband speech, and audio bit rate compressions in audiovisual communications. These signal classes differ in bandwidth, dynamic range, and in listener expectation of offered quality. It will become obvious that the use of our knowledge of auditory perception helps minimizing perception of coding artifacts and leads to efficient low bit rate coding algorithms which can achieve substantially more compression than was thought possible only a few years ago. The paper concentrates on worldwide source coding standards beneficial for consumers, service providers, and manufacturers     

Wideband speech and audio coding

Typical parameters of wideband speech and audio signals, including digitized versions of each, potential applications, and available transmission media, are described. Facts about human auditory perception that are exploited in audio coding and quality measures that play an important role in coder evaluations and designs are reviewed. Techniques for efficient coding of wideband speech and audio signals, with an emphasis on existing standards, are discussed. The audio coding standard developed by the Moving Pictures Expert Group within the International Organization for standardization (ISO/MPEG) is covered in some detail, since it will be used in many application areas, including digital storage, transmission, and broadcasting of audio-only signals and audiovisual applications such as videotelephony, videoconferencing, and TV broadcasting. Ongoing research and standardization work is outlined     

Bit rates in audio source coding

The goal is to introduce and solve the audio coding optimization problem. Psychoacoustic results such as masking and excitation pattern models are combined with results from rate distortion theory to formulate the audio coding optimization problem. The solution of the audio optimization problem is a masked error spectrum, prescribing how quantization noise must be distributed over the audio spectrum to obtain a minimal bit rate and an inaudible coding errors. This result cannot only be used to estimate performance bounds, but can also be directly applied in audio coding systems. Subband coding applications to magnetic recording and transmission are discussed in some detail. Performance bounds for this type of subband coding system are derived