小波变换音频编码算法研究

任意能量有限信号都可以用紧支撑正交小波基展开或分解，这一点对研究快速高效音频编码算法是非常重要的。本文设计一种基于正交小波变换的高保真音频编码算法，该算法可以把速率为７０５．６ｋｂｉｔ／ｓ的高保真音频信号压缩到１９２ｋｂｉｔ／ｓ，１６０ｋｂｉｔ／ｓ，１２８ｋｂｉｔ／ｓ，９６ｋｂｉｔ／ｓ和６４ｋｂｉｔ／ｓ，并保持重构音频信号的高质量。     

H.263: 极低码率窄带电信信道视频编码

介绍了码率低于６４ｋｂｉｔ／ｓ的窄带电信信道视频编码Ｈ．２６３建议的信源格式和编码算法。指出了极低码率视频编码技术不仅应用于公众电话网的可视电话传输，并且适合于移动环境，因而在多媒体电子邮件、交互式多媒体数据库、聋哑人视频辅助通信等领域得到了广泛的应用。     

低码率小波变换语音编码算法

本文给出了一种基于小波变换的低码率语音编码算法，它具有码率低（小于３ｋｂｉｔ／ｓ）、适用范围广、音质好、无人工噪声等优点，不仅对语音信号而且对音乐信号也有效，有助于实现低码率宽带声音编码，对信息的高效传输和存储有重要意义。     

交错—扩散卷积码的硬件实现及其性能

在现代数字通信系统中，纠错编码与交错技术相组合是系统抗突发干扰和随机干扰的有效措施。本文介绍了一种交错－扩散卷积码的原理及实现方法，详细讨论了交错及系统的硬件实现，并给出了实验结果，本系统在１０２４ｋｂｉｔ／ｓ信道上，在有一定保护长度的条件下，可纠正连续３００比特的突发误码；对随机误码改善可从１０＾－３改善到１０＾－７。对有突发和随机误码的混合信道具有较强的适应能力。     

基于队列置乱的小波域盲水印算法

随着Internet的普及和多媒体技术的发展，多媒体信息的安全及版权问题引起了人们越来越多的关注，数字水印技术应运而生，他通过向多媒体数据内隐藏版权信息来保护版权。首先，采用了一种新的基于队列变换的图像置乱技术，提高了水印抗剪切能力，并提供密钥增加了系统的安全性；其次，提出了一种新的基于小波域的盲水印算法，利用小波变换后的系数序列与给定序列之间的相关性来判断提取水印，提取无需原图像，是一种有效的盲水印算法。     

速率为64kbit／s—1920kbit／s会议电视的帧结构

本文主要介绍速率为６４ｋｂｉｔ／ｓ－１９２０ｋｂｉｔ／ｓ会议电视在Ｎ－ＩＳＤＮ信道和ＰＣＭ一次群信道上传输的各种帧结构。     

语音编码综述

语音编码经历了半个多世纪，其编码速度已从６４ｋｂｉｔ／ｓ压缩到８ｋｂｉｔ／ｓ，并向更低速率进展。本文将回顾这一历程，并探讨今后语音编码的研究方向。     

基于DSP的多进制正交码扩频系统

本文介绍了一种基于ＤＳＰＴＭＳ３２０Ｃ２５的多进制正交码扩频系统，该系统可以对２．４ｋｂｉｔ／ｓ和１６ｋｂｉｔ／ｓ数据进行扩频。文中讨论了其实现的关键技术。     

10km动力电缆高速数字信道的实现

介绍１０ｋｍ、１ｋＶ动力电缆上下行７６８ｋｂｉｔ／ｓ和上行３ｋｂｉｔ／ｓ非对称高速数字信道的实现。该信道用于上海交通大学和大洋委员会６０００ｍ深海拖曳系统中,用于传递实时图像和控制信号,经鉴定达到国际领先水平。介绍了系统构成,讨论了调制方式的选择、发送和接收滤波器的设计以及功率放大器的设计问题。最后指出了该项目的现实意义和应用前景。     

说说第三代移动通信系统

一、第三代移动通信系统概述 移动通信发展的最终目标是实现任何人可以在任何地点,任何时间与其他任何人进行任何方式的通信。到目前为止,移动通信系统已由第一代的模拟语音移动通信系统发展到第二代。第二代移动通信系统的业务种类主要限于话音和低速数据（≤９．６ｋｂｉｔ／ｓ）。 未来第三代移动通信是覆盖全球的多媒体移动通信,主要特征是可提供移动多媒体业务,其中高速移动环境支持１４４ｋｂｉｔ／ｓ,步行慢速移动环境支持３８４ｋｂｉｔ／ｓ,室内支持２Ｍｂｉｔ／ｓ的数据传输。其设计目标是为了提供比第二代系统更大的系统容…     

A speech coder for PC multimedia net‐to‐net communication

The paper presents a speech coding algorithm for operation at 11025 samples/s. The coder provides improved speech quality and compatibility with the MS‐Windows multimedia environment. The coding algorithm has been developed by adapting the ITU G729 and enhancing it with some recent developments in the medium band coding. The coder operates over a band of frequencies ranging from 20 to 5400 Hz at a bit rate of 8.9 kbit/s. Application of this coder includes intranet VoIP, voice chatting, multimedia communications, and voice archiving. Copyright © 2001 John Wiley & Sons, Ltd.     

Subjective Evaluation of Several Efficient Speech Coders

The voice quality of several 9.6 - 32 kbit/s coders is determined with an extensive set of subjective listening tests. Single encodings of μ255 PCM, adaptive differential PCM (ADPCM), subband coding (SBC), vocoder-driven adaptive transform coding (ATC), adaptive predictive coding (APC), and time domain harmonic scaling combined with SBC are compared in an idealized situation, that is, no added impairments. It is shown that single encodings of modest complexity 32 kbit/s coders such as ADPCM and SBC and more complex 24 kbit/s coders such as vocoder-driven ATC and APC offer quality nearly equivalent to 64 kbit/s μ255 PCM. However, these conclusions are drawn in the absence of a realistic telephone network where tandem encodings, delay limitations, and nonvoice signals exist. Tandem encodings of 64 kbit/s μ255 PCM, 32 kbit/s ADPCM, 16 kbit/s SBC, and 16 kbit/s APC are also evaluated. These 32 kbit/s and 16 kbit/s coders offer degraded tandem performance as compared to 64 kbit/s PCM, with the exception of synchronous tandeming of 32 kbit/s ADPCM with 64 kbit/s PCM where several encodings are subjectively equivalent to a single encoding of 32 kbit/s ADPCM.     

Synchronous Tandem Algorithm for 32-kbit/s ADPCM

The synchronous tandem property of nonaccumulation of distortion in tandem-connected ADPCM coders with a 64 kbit/s PCM interface is discussed here. The synchronous tandem algorithm used to provide this property in the steady-state mode is described with a case-by-case analysis, so as to show how the synchronous tandem property is realized in an ADPCM coder. A 32 kbit/s ADPCM coder utilizing this algorithm has been standardized by the CCITT (International Telegraph and Telephone Consultative Committee). The synchronous tandem property of the 32 kbit/s ADPCM coder is of great interest in network applications, because the ADPCM coder appears likely to be introduced into digital networks built partially with existing 64 kbit/s PCM circuits.     

基于小波变换的2.4kbit/s波形内插语音编码算法

基于双正交小波滤波器组对波形内插编码中提取的特征波进行多级分解与重构，提出了一种基于小波变换（WT）的2．4kbit／s特征波形内插（CWI）语音编码算法。编码端去除了特征波对齐运算，并对幅度谱进行多级分解，相位谱不传输，鉴于小波变换对信号的压缩特性，仅传输对人耳感知起主要贡献的最后一级特征波幅度谱；解码端对各尺度空间采用单独重建的方法，相位信息在重构的末级与幅度谱结合，并由浊音度标志选择固定或随机相位。此外，根据语音信号的时变特性，由基于子帧的浊音度标志选择需要传输的幅度谱及量化模式。主观R-A／B测试表明，这种基于小波变换的2．4kbit／s编码算法的合成语音质量明显优于标准的2．4kbit／s的MELP编码器及FS1016的4．8kbit／sCELP编码器，亦优于3．8kbit／s的传统CWI编码框架下的合成语音效果。     

Low-power and high-quality signal transmission baseband LSIC forpersonal communications

This paper presents a very low power consumption one-chip baseband large-scale integrated circuit (LSIC) for personal communication terminals. It comprises a π/4-shift QPSK modem, an adaptive differential pulse code modulation (ADPCM) codec, a time division multiple access time division duplex (TDMA-TDD) controller and a link access procedure for a digital cordless (LAPDC: Layer-2 protocol) controller. The developed LSIC meets all the specifications of the personal handy-phone system (PHS) standard. By employing a novel coherent demodulator and an ADPCM codec with a click noise suppressor, a higher quality voice transmission has been achieved in a fading environment. In addition, it has 61-kb/s data transmission capability to achieve wireless multimedia services based on PHS. Moreover, the novel circuit configurations of the modem, the ADPCM codec, the TDMA-TDD controller, and the LAPDC controller achieve significant power reduction of the baseband circuits (57.4 mW) of personal communication terminals. It enables very low power consumption wireless multimedia terminals to be achieved based on the PHS common air interface     

G.729协议在TMS320C6211上的实现

ITU—T提出基于CS—ACELP算法传输速率为8kbit／s的G．729协议，该协议可以用作如PHS（个人手持电话系统）、比特率低于64kbit／s的可视会议等多媒体服务，也可以作为有效利用传输能力的多路复用器。概要介绍了G．729协议的算法结构，对G．729协议在TMS320C6211DSP上的实现进行了研究。     

Variable rate speech coding for asynchronous transfer mode

A new speech coding and multiplexing scheme matched to the asynchronous transfer mode is described. A block coding technique that is based on a variable-rate coding algorithm that makes the most of the burstiness of voice information is employed. The main feature of the scheme is considerable bit reduction, which is attained by a fairly simple algorithm. It is demonstrated that the proposed algorithm exhibits better quality than that of a 32 kb/s ADPCM at a mean bit rate of less than 13 kb/s. The effect of statistical multiplexing is verified by means of simulation employing long conversational speech samples. Methods for constructing variable- and fixed-length frames (units of information multiplexed and transferred in the network) are proposed. The proposed coding algorithm is shown to be applicable to both variable- and fixed-length frame strategies     

基于TMS320DM6446的MELP/CELP混合编码方法研究与实现

在讨论TMS320DM6446结构的基础上,讨论了混合激励线性预测(MELP)的算法模型,并详细研究了MELP的几种关键技术.在MELP基础上,提出了MELP/CELP两种适用语音帧类型互补的编码方案相结合的语音编码方法,在周期性不强的语音帧采用CELP来弥补MELP的不足,并给出了这种编码器的结构和量化方案.使用TMS320DM6446实现发现可以获得与32kbit/s编码速率的ADPCM相近的语音质量.