ATM网络中语音编码和传输的新方案

本文针对未来新的ＡＴＭ通信方式,提出了一种新的语音可变速率编码和可变时延传输系统方案,为了将信号源和人耳听觉的特征,与ＡＴＭ网络的统计复用性相结合,实现语音的码率在缩和低时延传输,该方案将ＡＴＭ网络环境和语音编码系统中最优信号分析区间的选取、编码系统参数的确定相结合。文中基于一种新的分布熵进行信号特征判断,对输入信号构成不同的处理系统,具体编码由小波变换分带、多带二进树ＶＱ构成,输出码率可调,改变     

Voice packetization and compression in broadband ATM networks

Some methods of supporting voice in broadband ISDN, (B-ISDN) asynchronous transfer mode (ATM), including voice compression, are examined. Techniques for voice compression with variable-length packet format at DS1 transmission rate, e.g., wideband packet technology (WPT), have been successfully implemented utilizing embedded adaptive differential pulse code modulation (ADPCM) coding, digital speech interpolation (DSI), and block-dropping schemes. For supporting voice in B-ISDN, voice compression techniques are considered that are similar to those used in WPT but with different packetization and congestion control methods designed for the fixed-length ATM protocol at high speeds. Possible approaches for packetization and implementation of variable-bit-rate voice coding schemes are described. ADPCM and DSI for voice coding and compression and cell discarding (CD) for congestion control are considered. The advantages of voice compression and CD in broadband ATM networks are demonstrated in terms of transmission bandwidth savings and resiliency of the network during congestion     

Performance analysis of an ATM multiplexer loaded with VBR trafficgenerated by multimode speech coders

Multimode coders are able to exploit the different characteristics of the speech waveform and to take into account the different peculiarities of background noise, thus allowing improvements in both signal reconstruction and network-offered load. In this context the variable rate code excited linear prediction (VR-CELP) coding, that is, a multimode variable bit rate (VBR) coding based on the CELP technique, has been introduced in the literature and is currently being considered for use in various applications, especially in the third-generation UMTS cellular systems. The target of the paper is to introduce an efficient and accurate framework allowing a network designer to analyze the impact of multimode VBR speech coding on the quality of service (QoS) provided by a wireless/wired ATM network. In order to capture the coder output characteristics, we propose to model a VR-CELP voice source by using a switched batch Bernoulli process (SBBP). More specifically, three models are introduced and compared in terms of accuracy and simplicity in determining network performance. As a result of the comparison, a four-state model has been chosen as the best tradeoff. The model is then used to analytically derive the loss probability and the jitter probability density function of an ATM multiplexer loaded by a number of VR-CELP sources. Finally, the proposed paradigm has been assessed in a case study where we demonstrate that, for a given output ATM link capacity and for a number of telecommunication services involving voice transmission, VR-CELP coding performs better than traditional on-off coding     

截幅失真对低速语音编码的影响分析

截幅失真会影响语音编码质量,特别在低速率语音编码条件下,截幅语音不再符合人发声模型,编码语音质量严重下降。为了研究截幅失真对低速率语音编码的影响,从截幅语音编码参数提取和截幅语音编码质量两个方面进行了分析。采用偏离度衡量低速率语音编码参数提取的准确性,编码参数包括LPC、基音周期和清浊音。在不同截幅程度下,分析了各种参数错误分布、错误类型和错误原因。采用客观感知语音质量评估PESQ打分评估截幅语音编码质量。针对常用截幅修复方法在截幅程度较大时修复性能下降严重的现象,提出采用两种改进型截幅修复方法对截幅语音进行修复。实验结果表明,改进的截幅修复方法能有效提高截幅程度较大时的低速语音编码质量。      

Variable bit-rate coding of video signals for ATM networks

Statistical characteristics of video signals for video packet coding, are clarified and a variable-bit-rate coding method for asynchronous transfer mode (ATM) networks is described that is capable of compensating for packet loss. ATM capabilities are shown to be greatly affected by delay, delay jitter, and packet loss probability. Packet loss has the greatest influence on picture quality. Packets may be lost either due to random bit error in a cell header or to network control when traffic is congested. A layered coding technique using discrete-cosine transform (DCT) coding is presented which is suitable for packet loss compensation. The influence of packet loss on picture quality is discussed, and decoded pictures with packet loss are shown. The proposed algorithm was verified by computer simulations     

A New Speech Codec Based on ANN with Low Delay

The author designs a new speech codec in this paper, which is based on ANN to carry out nonlinear prediction. This new codec synthesizes speeches with better quality than the conventional waveform or hybrid codecs does at the same bit rate. Moreover, the most important characteristic of this codec is the low coding delay, which will benefit the enhancement of the speech communication QoS when we transmit speech signals in IP or ATM networks.     

Missing packet recovery techniques for low-bit-rate coded speech

Since missing-packet recovery techniques for conventional PCM speech are not applicable to packetized speech communication systems with low-bit-rate coding schemes, quality degradation mechanisms are presented for missing-packet recovery techniques. These mechanisms are least significant bit (LSB) dropping, waveform substitution, and odd-even sample-interpolation schemes. A comparison of these techniques in terms of signal-to-noise ratio and perceptual distortion under packet loss conditions shows that the LSB-dropping scheme with embedded coding is the most promising technique for recovering missing packets     

语音业务在ATM网络中降低装配时延提高系统质量的新方法

本文针对ＡＴＭ网络中的语音通信，提出了一种降低信元装配时延的新方法，以及相应的基于ＡＤＰＣＭ的新的信源编码系统，它能确保在信元丢失时，收发双方编译码系统不会失步，并且能产生基本无附加听觉噪声的丢失段恢复语音。文中还分析了各部分参数的求法，实验表明，该法产生的丢失段恢复语音，具有良好的听觉质量。     

一种适用于ATM的语音处理方案

以ＡＴＭ实现分组话音通信，需要解决两个基本问题：传输时延和分组丢失。针对这两个基本问题，本文着重讨论了ＡＴＭ分组话音通信中３２ｋｂ／ｓＡＤＰＣＭ话音分组丢失的重建技术－－于模式匹配的波形替代技术和静默重建技术，分别用以补偿由于网络阻塞造成的分组话音信息丢失而产生的失真和改善重建话音的自然性。     

Advances in speech and audio compression

Speech and audio compression has advanced rapidly in recent years spurred on by cost-effective digital technology and diverse commercial applications. Recent activity in speech compression is dominated by research and development of a family of techniques commonly described as code-excited linear prediction (CELP) coding. These algorithms exploit models of speech production and auditory perception and offer a quality versus bit rate tradeoff that significantly exceeds most prior compression techniques for rates in the range of 4 to 16 kb/s. Techniques have also been emerging in recent years that offer enhanced quality in the neighborhood of 2.4 kb/s over traditional vocoder methods. Wideband audio compression is generally aimed at a quality that is nearly indistinguishable from consumer compact-disc audio. Subband and transform coding methods combined with sophisticated perceptual coding techniques dominate in this arena with nearly transparent quality achieved at bit rates in the neighborhood of 128 kb/s per channel     

IP和ATM网络中语音分组丢弃对解码语音质量的影响及对应的措施

基于分组传输的网络(如IP和ATM)，进行实时语音通信时，存在分组可能丢失，因而影响通信质量的问题。本文提出两种新的方法，以减少或降低语音分组丢失带来的影响。实验结果表明，它们明显改善了语音通信质量。     

数字集群通信系统中的变速率语音编码

变速率语音编码技术对于提高移动通信系统的话音服务质量起着十分关键的作用。论文首先介绍变速率语音编码技术的基本原理和相关技术,然后阐述数字集群通信系统中所采用的变速率语音编码方法,最后展望了数字集群领域变速率语音编码的发展方向。     

Turbo乘积码在卫星ATM系统中的应用

Turbo乘积码(TPC)是一种性能优异的前向纠错编码技术.卫星ATM技术代表了卫星通信网络的研究方向.研究了TPC在卫星ATM系统中的应用方案.首先,分析了卫星信道对ATM信元传输带来的影响;然后对TPC的编、解码的方法进行了简要介绍,并阐述了TPC的优越性能;最后,提出了一种卫星ATM网络中应用TPC的解决方案,并通过仿真得到了该方案的性能曲线.仿真结果证明,TPC的应用能够大幅降低卫星信道的误码率,使ATM信元能够正确传输.     

A New Method of Reducing Assembly Delay of Speech Packets

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ATM网络中电路仿真业务定时恢复的改进方法

ＡＴＭ技术很好地解决了数据的透明传送问题，但定量信息在ＡＴＭ网中的传递透明性由于统计复用而受到损害，目前用于电路仿真精力的定量恢复方法根据是否使用公共时钟大致分为同步和异步两种模式，由于ＳＤＨ网的推广，同步方式已ＩＴＵ－Ｔ采用，主要有ＳＦＥＴ、ＴＳ和ＳＲＴＳ三种，由于ＡＴＭ业务的多样性，它们最大都有可能引入ＩＵＩ的低频抖动，本文提出了一种改进方法，首次将２级∑－△量化应用于定量恢复，可使任何业务的     

The impact of the ATM concept on video coding

The induction of the asynchronous transfer mode (ATM) concept may significantly influence the coding of video services for broadband networks. The authors show how the absence of a physical channel structure and the ability to switch bursty traffic can be used to enhance video coding. Packetization defects and their impact on picture quality, coding algorithms, and synchronization schemes are studied. The authors describe variable-bit-rate coding and report on the results obtained with a hardware implementation of a variable-bit-rate video codec. Statistical multiplexing gain figures are given. The influence of cell loss on image quality is discussed and simulation results are given. A layered coding model offering good cell loss concealment properties and high flexibility is described     

Vocoders: Analysis and synthesis of speech

Techniques for analysis and synthesis of speech signals are reviewed with emphasis on vocoders and related devices for more efficient transmission and storage of speech. Selected applications of speech coding methods as sensory aids to the handicapped are described.     

CDMA的两种变速率语音编码性能比较

从语音编码器的编码原理出发分别介绍了CDMA的两种变速率语音编码标准———EVRC和SMV,并在此基础上进行了性能分析与比较,最后得出比较结果:SMV工作在模式0的时候,速率与EVRC相当,但其合成音质有明显提高;SMV相较EVRC的优点还体现在他的灵活性,编码速率和模式可变,能够使通话质量与系统容量达到不同程度的均衡。     

Speech coding in mobile radio communications

Speech coding, the efficient representation of speech in digital form, is one of the key technologies in current and evolving digital cellular and wireless voice communications offerings. The speech coders in existing standards exhibit a level of sophistication and performance unimaginable just 15 years ago. We outline the characteristics of the mobile communications problem with respect to speech coders and point out the principal issues in speech coder design for these applications. Speech coding methods in existing mobile communications standards are described and contrasted. The limitations imposed by the wireless channel and by background impairments are discussed, and approaches to addressing their resulting effects are presented. Suggestions for future research in speech coding for the mobile communications problem are outlined     

基于VAM的AAL1实时话音接入

提出一种基于ＶＡＭ的实时语音接入方法，它利用ＡＴＭ的统计复用特性，通过将多路语音复用在一个ＡＡＬ１上，从而降低单路话音的装配时延，可以在不降低通话质量的情况下保留ＡＴＭ的有效带宽。实验表明，这种方案可以得到很好的效果。