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。     

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     

G.723.1 Codec在OpenH323协议栈上的实现

OpenH323是一个开放源码的VoIP(Voice over IP)协议栈,支持H．323和SIP等多媒体通信协议,为多媒体应用提供了一个很好的开发平台。G．723．1是ITU-T建议在中低速率多媒体通信中使用的语音压缩算法,目前该算法已在IP电话系统中得到广泛应啊。基于OpenH323协议栈实现G．723．1Codec有着十分重要的应用价值。介绍在OpenH323的软件终端上实现G．723．1Codec的基本方法,并可推广到G．729等其它多种语音压缩算法。     

ITU-T G.729 Annex A: reduced complexity 8 kb/s CS-ACELP codec fordigital simultaneous voice and data

This article describes the ITU-T Recommendation G.729 Annex A (G.729A) for encoding speech signals at 8 kb/s with low complexity. G.729A is the standard speech coding algorithm for multimedia digital simultaneous voice and data (DSVD). G.729A is bitstream interoperable with G.729; that is, speech coded with G.729A can be decoded with G.729, and vice versa. Like G.729, it uses the conjugate-structure algebraic code excited linear prediction (CS-ACELP) algorithm with 10 ms frames. However, several algorithmic changes have been introduced which result in a 50 percent reduction in complexity. This article describes the algorithm introduced to achieve the low complexity goal while meeting the terms of reference. Subjective tests showed that the performance of G.729A is equivalent to both G.729 and G.726 at 32 kb/s in most operating conditions; however, it is slightly worse in the case of three tandems and in the presence of background noise. A breakdown of the complexities of both G.729 and G.729A is also given     

ITU-T Coders for Wideband, Superwideband, and Fullband Speech Communication

This section of the magazine presents recent algorithms developed by the ITU to provide high quality coding beyond traditional narrowband telephony. Speech coders can be characterized by their bit rate, quality, complexity, and delay. Typical applications fall into one of two categories, one-way and two-way. The first includes storage applications such as telephone answering systems, streaming, multimedia delivery, and push-to-talk calls. The second includes realtime communications such as two person phone calls and conference calls. In this latter category, if the delay is too large - exceeding 300 ms round-trip - humans have difficulty communicating, while for storage and playback operations delay is not a factor. The complexity of a speech coder is one of the main contributing factors to its cost and energy usage. Complexity is most often measured in terms of memory usage (both RAM and ROM) and the number of instructions executed per second. All applications are sensitive to cost, and many are sensitive to energy usage as well. The desired bit rate is determined by channel capacity or storage capacity, depending on the application.     

基于VC5409的双速率语音编译码器的实现

ITU-T.G.723.1为国际电信联盟(ITU)制定的5·3bit/s和6.3kbit/s双速率语音编码建议,分别采用代数码激励线性预测(ACELP)算法和多脉冲最大似然量化(MP-MLQ)算法。在阐述G.723.1建议编译码算法的原理和实现的基础上,重点介绍了在开发基于TMS320VC5409实时实现该建议的全双工编译器过程中所做的工作。该语音编译码器通过了G.723.1所有测试矢量的验证。     

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.     

ITU G.732.1语音编码器在DSP上的实时实现

ＩＴＵＧ。７２３．１建议的编码速度为５．３和６．３ｂｉｔ／ｓ的双码率编码方案是一种运算复杂的用于多媒体通信中的低码率语音编码器。本文介绍了该编码器的算法原理和在一片ＴＭＳ３２０Ｃ５４１定点ＤＳＰ芯片上实时实现该编码器过程中的软，硬件结构及关键技术。     

Integration of harmonic and analysis by synthesis coders

Hybrid coding of speech has been proposed to overcome the limitations of a single model in representing the wide variety of characteristics of human speech. A new hybrid coding algorithm, which combines harmonic and analysis by synthesis coding techniques, is presented. To integrate the harmonic and analysis by synthesis coders, novel phase synchronisation and speech classification techniques are developed. The perceptual quality of the speech synthesised using the unquantised hybrid model is almost indistinguishable when compared with 128 kbit/s linear PCM. Two variable rate coders are developed based on the designed hybrid model, by quantising the parameters at different bit rates. Subjective listening tests show that the speech quality of the variable rate hybrid coders outperform the quality of 5.3 kbit/s and 6.3 kbit/s ITU G.723.1 coders, at maximum bit rates of 4 kbit/s and 6 kbit/s respectively.     

Channel and source considerations of a bit-rate reduction technique for a possible wireless communications system's performance enhancement

In wireless commercial and military communications systems, where bandwidth is at a premium, robust low-bit-rate speech coders are essential. They operate at fix bit rates and those bit rates cannot be altered without major modifications in the vocoder design. A novel approach to vocoders, in order to reduce the bit rate required to transmit speech signal, is proposed. While traditional low-bit-rate vocoders code original input speech, the proposed procedure operates on the time-scale modified signal. The proposed method offers any bit rate from 2400 b/s to downwards without modifying the principle vocoder structure, which is the new NATO standard, Stanag 4591, Mixed Excitation Linear Prediction (MELP) vocoder. We consider the application of transmitting MELP-encoded speech over noisy communication channels by applying different modulation techniques, after time-scale compression is applied. Three different time-scale modification algorithms have been evaluated and waveform similarity overlap and add (WSOLA) algorithm has been selected for time-scale modification purposes. Computer simulation results, both source and channel, are presented in terms of objective speech quality metrics and informal subjective listening tests. Design parameters such as codec complexity and delay are also investigated. Simulation results lead to a possible wireless communications system, whose performance might be enhanced by using the spared bits offered by the procedure.     

iLBC语音编解码器核心技术的研究

重点讨论了iLBC编解码器独立于帧的长期预测。独立于帧的长期预测是用来在编码语音没有遭受与传输丢失相关的多帧语音退化情况下,开发斜度标记相关的办法。然后介绍了iLBC,G.729A和G.723.1编解码器的平均主观得分MOS,并用信号为例说明基于独立于帧的长期预测编解码器和CELP编解码器之间的不同,最后用语音重构的例子说明二者语音质量间的差别。     

基于多媒体处理器TM1000语音编解码器的实时实现

Ｇ．７２３．１协议是ＩＴＵ－Ｔ（国际电信联盟）为多媒体通信制定的双速率语音编码的国际标准。ＴＭ１０００是一种带有ＶＬＩＷ（超长指令字）结构的多媒体处理器,可应用在可视电话,会议电视,ＪＰＥＧ,ＭＰＥＧ－１等通信领域。介绍一种采用ＴＭ１０００处理器平台实时实现符合Ｇ．７２３．１协议的语音编解码器。     

ITU-T G.729.1 Scalable Codec for New Wideband Services

G.729.1 is a scalable codec for narrowband and wideband conversational applications standardized by ITU-T Study Group 16. The motivation for the standardization work was to meet the new challenges of VoIP in terms of quality of service and efficiency in networks, in particular regarding the strategic rollout of wideband service. G.729.1 was designed to allow smooth transition from narrowband (300-3400 Hz) PSTN to high-quality wideband (50-7000 Hz) telephony by preserving backward compatibility with the widely deployed G.729 codec. The scalable structure allows gradual quality increase with bit rate. A low-delay mode makes the coder especially suitable for high-quality speech communication. The article presents the standardization goals and process, an overview of the coding algorithm, and the codec performance in various conditions.     

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     

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。     

Candidate scheme for fast ACELP search

The algebraic code excited linear prediction (ACELP) algorithm has been adopted by many speech coding standards, due to low complexity and high quality in its analysis-by-synthesis optimisation. For further computational complexity reduction, the authors propose a fast ACELP algorithm using a designed pilot function to predict the predetermined candidate pulses. With candidate pulses, it is possible to not only reduce the number of search loops but also to avoid the computation of unnecessary correlation functions. The proposed candidate position scheme can be applied to all ACELP coders such as the ITU G-723.1 and G.729 as well as the GSM enhanced full rate (EFR) speech coding standards. Simulation results show that the computational load can be reduced by about 50-80% with almost imperceptible degradation in performance     

Low delay coder (< 25 ms) of wideband audio (20 Hz-15 kHz) scalable from 64 to 32 kbit/s

A low delay coder for speech and music signals sampled at 32kHz is described. Its algorithmic delay does not exceed 25 ms which enables audioconferencing applications without echo cancellation. Its bit rate is scalable between 64 and 32 kbit/s by steps of 8 kbit/s. The transmitter issues the binary code at 64 kbit/s with lower bit rate codes embedded in it. The receiver may operate at lower bit rates with gradual loss of quality. The proposed coder is based on a mixed scheme : the adopted solution contains elements from the CELP speech coder and frequency domain music coders. The perceptual signal is obtained in the time domain, then transformed to the frequency domain where bit allocation is calculated and transform coefficients are quantized. A first solution based on the dft is discussed, then a second solution based on a mdct with small overlap is applied. The quantization of these coefficients is done in the following way. First, a prediction of the whole spectrum is applied. Then, a mean- removed gain- shape split vq is used for amplitude spectrum quantization and a hierarchical 2- dimensional vq is used for phase spectrum quantization with amplitude correction. At the phase quantization stage, each codeword describing the selected vector index is split into parts corresponding to different bit rates. Due to the hierarchical codebook structure, truncated indices may be used, without much affecting the signal quality. Simulation results are presented and the robustness of the proposed coder is examined.     

The H.324 multimedia communication standard

ITU-T H.324 is the new international standard for multimedia communication on low-bit-rate circuit-switched networks, including ordinary analog telephone lines. It supplies any combination of real-time video, audio, and data. H.324 was designed, above all, to provide the best possible performance (video and audio quality, delay, etc.) on low-bit-rate networks. This article covers the H.324 system and its component standards, including the V.34 modem, H.223 multiplexer, H.245 control protocol, G.723.1 audio coder, and H.263 video coder. Call setup procedures and optional features like encryption and data application protocols are also described. The standard should ensure interoperability among a diverse variety of H.324 terminals, including PC-based multimedia videoconferencing systems, voice/data modems, encrypted telephones, World Wide Web browsers with live video. Remote security cameras. and standalone videophones. Products compliant with H.324 should all interoperate with each other and, through appropriate gateways, with ITU-T H-series terminals on ISDN, LANs, and ATM/B-ISDN networks     

Evaluation of six medium bit-rate coders for the Pan-Europeandigital mobile radio system

A digital cellular mobile radio system has been under development in Europe since 1982 under the coordination of the working group CEPT GSM (groupe speciale mobile). In a recent coordinated experiment, listening opinion tests were performed on the speech output of six candidate 16 kb/s speech coding schemes for this system: one regular-pulse excited coder, one multiple-excited coder, and four subband coders. For comparison purposes, test conditions from a companded cellular FM system currently in operation were included in the experiment. The six codecs were companded in terms of subjective quality, transmission delay, and ease of implementation. In this overall comparison, no single codec was superior in all respects. However, the regular-phase-excited linear predictive coder, which provided the best speech quality, had acceptable complexity and delay and was singled out for further improvement. Ultimately, an improved version of this codec, a regular-pulse-excited/long-term-prediction LPC coder was selected     

采用Trimedia/Nexperia DSP的流结构模式实现G.723.1实时语音编解码

G．723．1协议是将音频信号压缩到5．3Kb／s和6．3Kb／s2种码率的音频编解码协议，主要应用在多媒体通信中的音频压缩。介绍一种基于Trimedia／Nexperia DSP的流结构TSSA(Trimedia Streaming Software Architecture)实现应用在可视电话中的G．723．1实时语音编解码。