Low bit-rate speech coders for multimedia communication

The International Telecommunications Union (ITU) has standardized three speech coders which are applicable to low-bit-rate multimedia communications. ITU Rec. G.729 8 kb/s CS-ACELP has a 15 ms algorithmic codec delay and provides network-quality speech. It was originally designed for wireless applications, but is applicable to multimedia communications as well. Annex A of Rec. G.729 is a reduced-complexity version of the CS-ACELP coder. It was designed explicitly for simultaneous voice and data applications that are prevalent in low-bit-rate multimedia communications. These two coders use the same bitstream format and can interoperate. The ITU Rec. G.723.1 6.3 and 5.3 kb/s speech coder for multimedia communications was designed originally for low-bit-rate videophones. Its frame size of 30 ms and one-way algorithmic codec delay of 37.5 ms allow for a further reduction in bit rate compared to the G.729 coder. In applications where low delay is important, the delay of G.723.1 may be too large. However, if the delay is acceptable, G.723.1 provides a lower-complexity alternative to G.729 at the expense of a slight degradation in quality. This article describes the attributes of speech coders such as bit rate, complexity, delay, and quality. Then it discusses the basic concepts of the three new ITU coders by comparing their specific attributes. The second part of this article describes the standardization process for each of these coders     

高质量4kb/s FS-ACELP语音编码算法及性能

4kb/s有限状态代数码激励线性预测语音编码算法FS-ACELP是一种具有延时较短、合成语音质量高、算法复杂度较低的语音编码算法.在线性预测(LP)参数量化上,利用了语音帧内和帧间的相关性,对线谱对(LSP)参数使用预测式分裂式矢量量化,获得很高的量化效率.在自适应码本搜索上,采用了有限状态控制分数延时搜索的算法,在保证合成语音质量的同时,有效地降低了运算量.对于随机码本,采用了具有多模结构的代数码本,提高语音合成质量.对于激励码序列的增益,采用了预测式矢量量化,有效地提高了量化精度.经非正式听音测试,4kb/s FS-ACELP的合成语音质量超过了北美8kb/s VSELP,接近G.729 8kb/s CS-ACELP,MOS分约为3.9.     

高质量4kb/s FS-ACELP语音编码算法及性能

4kb/s有限状态代数码激励线性预测语音编码算法FS-ACELP是一种具有延时较短,合成语音质量高、算法复杂度较低的语音编码算法.在线性预测(LP)参数量化上,利用了语音帧内和帧间的相关性,对线谱对(LSP)参数使用预测式分裂式矢量量化,获得很高的量化效率.在自适应码本搜索上,采用了有限状态控制分数延时搜索的算法.在保证合成语音质量的同时,有效地降低了运算量.对于随机码本,采用了具有多模结构的代数码本,提高语音合成质量.对于激励码序列的增益,采用了预测式矢量量化,有效地提高了量化精度.经非正式听音测试,4kb/s FS-ACELP的合成语音质量超过了北美8kb/s VSELP,接近G.729 8kb/s CS-ACELP,MOS分约为3.9.     

Three new speech coders from the ITU cover a range of applications

Many new speech coding standards have been created in the 10-year period 1987-1996. The author reviews the key attributes that determine what coder to select for different applications. The article then focuses on three new speech coding recommendations from the ITU-T, namely G.723.1, G.729, and Annex A of G.729. They provide good coverage for a wide range of applications that have low bit rate requirements (i.e., from 5.3 to 8 kb/s). In addition to bit rate, the article reviews their delay, complexity, and performance. Also reviewed are the history of these standards, and what considerations influenced the requirements each of these coders had to meet     

低比特率语声编码器的新发展

ITU-T建议G.729、G.729 AnnexA和G.723.1是国际电信联盟(ITU)最新颁布的3种适用于多媒体通信的低比特率线性预测语声编码器标准。文章介绍了语声编码器的比特率、复杂度、延迟和音质等性能指标的含义,并通过比较3种标准的新型声码器在算法和性能指标上的异同点,讨论了它们在多媒体通信中的不同应用。     

一种应用于音库压缩的低速率编解码方案

传统的语音编解码器大多针对通信场合设计,无法很好的满足应用于语音合成中的音库压缩的要求。本文根据音库压缩的特点,提出了一种低码速率的编解码方案,其在3Kbps左右的码率下获得类似G.723.1在5.3Kbps下的效果,并具有解码端算法简单的优点。作为技术原型,该方案可以广泛应用于各种场合下的语音合成系统,特别对资源受限情况下语音合成,具有重要的意义。     

The road to G.729: ITU 8-kb/s speech coding algorithm with wirelinequality

In November 1995 the International Telecommunication Union Telecommunications Sector (ITU-T) approved an 8-kb/s speech coding algorithm with wireline quality. This culminated the effort that the CCITT had set in motion in 1990. This article presents the methods for managing the project through its major milestones from setting the terms of reference to the selection, testing, optimization, and dissemination of the algorithm. While G.729 was being finalized, a new requirement for a low complexity 8-b/s speech coding arose. This article explains how the change in scope was accommodated without the unnecessary proliferation of incompatible algorithms     

G.729自适应码书搜索算法的研究

G.729语音编码算法复杂,很大程度上要归因于码书搜索算法。为了降低码书搜索复杂度,G.729的简化版G.729A采用了自适应码书的偶样点开环基音搜索,使得编码复杂度大为降低,不过编码仍要花费很多的时间。通过对G.729码书搜索算法的研究,提出了对自适应码书的改进。改进了自适应码书搜索的G.729编码语音质量不低于G.729A的语音质量,但自适应搜索复杂度大大降低。     

基于ARM SoC的G.729A编码实现

G.729A是国际电信联盟(ITU)于1996年提出的一种语音压缩标准,在G.729基础上进行了必要的简化,该算法是一种共轭结构代数码激励线性预测(CS-ACELP)的8 kbits语音编码算法.文中主要介绍一种基于ARM处理器的系统架构和G.729A编码的过程,以及将该编码算法应用于此系统需要注意的一些原则和方法.针对算法本身采用了一些快速算法,并且针对ARM处理器的特性进行了一些优化,以便能降低编码复杂度,做到实时编码.     

高质量4~8kb/s变速率有限状态ACELP语音编码算法研究

4~8kb/s变速率有限状态代数码激励线性预测语音编码(VR-FS-ACEL)是一种具有延时较短、合成语音质量高、算法复杂度较低的语音编码算法.在线性预测(LP)参数量化上,使用预测式分裂式矢量量化,获得很高的量化效率.在自适应码本搜索上,采用了有限状态控制分数延时搜索的算法,有效地降低了运算量.对于随机码本,采用了具有多模结构的代数码本,提高语音合成质量.对于激励码序列的增益,采用了预测式矢量量化,有效地提高了量化精度.经非正式听音测试,其中4kb/s的合成语音质量超过了北美8kb/s VSELP,接近长途质量,而6kb/s和8kb/s合成语音质量达到了长途质量,与G.7298kb/s CS-ACELP相当.     

从AMR到G.729的参数层编码转换方案

分析了Adaptive Multi Rate(简称AMR)编码方案与G．729编码方案的异同，在此基础上介绍了一种从AMR编码到G．729编码的参数层直接转换方案。与传统的编码转换方案相比，在语音质量的损失可以接受的前提下，算法复杂度有较大的降低。     

高质量的4 kb/s散布脉冲CELP语音编码算法

本文提出了一种散布脉冲CELP(DP-CELP)语音编码算法,激励矢量由特殊结构的代数码书与固定形式的散布脉冲的卷积获得,这种激励源有效地改善了重建语音质量,但未增加代数码书搜索的复杂度.非正式的主观听力测试表明,这种4 kb/s DP-CELP语音编码算法的合成语音质量非常接近G.723.1中6.3 kb/s语音编码器.     

Performance of ITU-T G.729 8 kb/s CS-ACELP speech codec withnonvoice narrowband signals

A report is given on the results of a series of objective measurements conducted by COMSAT in a laboratory environment aimed at characterizing the narrowband performance of the ITU-T G.729 8 kb/s conjugate-structure algebraic code-excited linear prediction (CS-ACELP) speech coder. The test procedures followed ITU-T Recommendation G.720, “Characterization of Low-Rate Voice Coder Performance with Non-Voice Signals”. It was concluded that the G.729 algorithm has excellent performance with narrowband signals in general (e.g., single tones and DTMF signals). As for Signaling System No. 5 (SS5) interregister signals, the G.729 CS-ACELP frequently failed to correctly identify SS5 digit 6 in a number of occurrences, using worst-case analysis equipment. This indicates that the SS5 performance of G.729 codecs in trunks where SS5 is used should be carefully measured before the network planner decides on its deployment. Great care should also be taken for tandem connections, since no test has been performed for these configurations     

Characterizing the subjective performance of the ITU-T 8 kb/sspeech coding algorithm-ITU-T G.729

Three listening-only experiments were conducted to characterize the subjective performance (i.e., speech quality) of 8 kb/s G.729. These experiments evaluated the quality of coded speech under a variety of conditions: (i) interworking with other international and regional speech coding standards; (ii) input speech that had been corrupted by environmental noise; (iii) operation over degraded transmission channels (including random bit errors and a simulated radio channel). The results of these experiments indicate that 8 kb/s G.729 meets the performance requirements that were established at the beginning of the standardization process     

低速率扩展编码标准G.729D

１９９８年 ９月,ＩＴＵ－Ｔ在 Ｇ．７２９的基础上制定了 Ｇ．７２９Ｄ。 Ｇ．７２９Ｄ是 Ｇ．７２９的低速率扩展标准,当速率为 ６．４ｋｂ／ Ｓ时仍能保持很好的话音质量。文章介绍了 Ｇ．７２９Ｄ基本原理,对Ｇ．７２９Ｄ与Ｇ．７２９进行了比较和分析,并给出了主观测试结果。     

ITU-T G.711.1: Extending G.711 to Higher-Quality Wideband Speech

In March 2008 the ITU-T approved a new wideband speech codec called ITU-T G.711.1. This Recommendation extends G.711, the most widely deployed speech codec, to 7 kHz audio bandwidth and is optimized for voice over IP applications. The most important feature of this codec is that the G.711.1 bitstream can be transcoded into a G.711 bitstream by simple truncation. G.711.1 operates at 64, 80, and 96 kb/s, and is designed to achieve very short delay and low complexity. ITU-T evaluation results show that the codec fulfils all the requirements defined in the terms of reference. This article presents the codec requirements and design constraints, describes how standardization was conducted, and reports on the codec performance and its initial deployment.     

基于G.729A的CS-ACELP语音压缩算法的改进

通过对CS-ACELP语音压缩算法的研究,分析了其各个模块复杂度,为使此算法更适合对硬件要求比较苛刻的系统,必须选择运算峰值比较高的模块进行改进。本文在CS-ACELP简化版本G.729A的基础上,主要针对复杂度比较高的自适应码本搜索模块进行改进,最后通过仿真实验,证明了改进算法的有效性。     

G.729AB算法在DM642上的优化实现

G.729AB是ITU提出的中低速语音编解码算法,具有语音质量高、低延时和稳定性好的优点。在研究算法基本原理的基础上,讨论了G.729AB在DSP芯片的优化实现方法,采用C和汇编混合语言实现了G.729AB语音编解码器算法。实验结果表明,优化后代码性能得到了很大提高,达到了优化实现的目的,可广泛运用于可视电话、IP电话、移动通信和公共电话交换网,具有一定的现实意义。     

一种基于G.729的CS-ACELP新算法

在G．729的基本框架下,码本搜索采用次最优部分码本快速搜索法,知觉加权滤波器使用基于声学心理模型的知觉加权滤波器,使得8Kb／s共扼结构代数码激励线性预测语音编码在不降低语音质量的情况下降低计算复杂度。次最优部分码本快速搜索方法虽然降低搜索准确度,但是大大减小了搜索时的数据处理量;声学心理滤波器考虑人耳对不同频率信号的不同敏感度,因而能获得更好的主观音质效果。仿真结果表明,该算法复杂度降低,并取得满意的合成语音质量。