Application of 32 and 16 kb/s speech encoding techniques to digital satellite communications

This paper describes the performance of various voice encoding techniques at 32 and 16 kb/s for applying to digital satellite communication systems. The subjective performances of adaptive differential PCM (ADPCM), adaptive predictive coding (APC), subband coding (SBC) and adaptive delta modulation (ADM) are compared under various satellite channel environments, that is, random and burst channel errors in satellite link and an ambient noise in the ship-to-shore direction in a maritime satellite channel. The performance of the voiceband data at 4·8 and 2·4 kb/s is also evaluated for these coders. ADPCM encoding at 32 kb/s is very attractive for conventional fixed satellite systems, keeping the equivalent quality to 64 kb/s PCM. On the other hand, APC encoding at 16 kb/s is also most suitable for maritime satellite communication systems at the sacrifice of a small degradation of speech quality.     

APC-AB codec modules operating at 16 and 8 kbit/s

High-quality speech codec modules operating at 16 and 8 kb/s have been developed using an adaptive predictive coding with adaptive bit allocation (APC-AB) scheme. An optimized APC-AB algorithm is studied that reduces processing complexity while maintaining speech quality. The coding algorithm is implemented in two digital signal processors (DSPs). The DSP chips, a framing LSI circuit, a PCM codec, and some peripheral ICs are integrated in each of two compact packages, i.e. codec modules, operating at 16 or 8 kb/s. The codec module size is as small as 80 mm×50 mm×12 mm, and its typical power consumption is 500 mW using 2-μm CMOS LSI technology. At 16 kb/s this APC-AB codec achieves high speech quality, close to that of a 7-bit μ-law PCM. The codec modules are expected to be used for various applications such as customer premises multiplexers for digital leased lines, digital mobile radio, and stored-and-forward-message systems (voice-mail systems)     

低速率语音编码技术

本文以ITU最新公布的语音编码方案MP MLQ /ACELP为例 ,介绍低速率语音编码技术的发展 ,并将该方案和GSM系统的全速率语音编码方案进行比较 ,说明中低速率语音编码在技术和应用上的一些特点。     

Operating aspects of new LRE/DSI DCMEs

Digital circuit multiplication equipment (DCME) that uses a combination of talk-spurt interpolation and low-rate encoding (32 kb/s ADPCM rather than the traditional 64 kb/s PCM) is about to be introduced in operating satellite systems. These DCMEs possess a number of important features that have been introduced to achieve a suitable balance between high channel multiplication ratios and high quality voice and data transmission. The methods used to accomplish this are generation of overload channels to virtually eliminate talk-spurt clipping, establishing a limit on the fraction of channels that undergo bit reduction by use of dynamic load control, automatic routeing of trunks carrying in-band data signals to non-interpolated bearer channels, use of a bit-bank approach to provide five-bits/sample ADPCM coding for in-band data channels and rotation of the server channels among the talk-spurts to spread the degradation caused by bit reduction uniformly across all talk-spurts. This paper analyses the performance of the DCME embodying the above features, and presents results in terms of the input-to-output channel multiplication ratio (CMR) as a function of the number of bearer channels (over a range from 4 to 61 bearer channels) and the number of input channels carrying in-band data. Extending the operation range down to as few as four bearer channels permits evaluation of the effectiveness of multiclique operation. In addition, the influence on CMR of (a) non-bit-bank operation, that would be possible if an ADPCM codec capable of supporting in-band data carriers at four-bits/sample were used, and (b) a short-hangover-time speech detector are examined. Use of hangover times less than 30 ms are shown to achieve a CMR of as great as 3.3: 1.     

语音数字信号压缩技术在TES系统中的应用

介绍ADPCM标准、RLPC编码原理,编、解码器方框图及工作过程。民航卫星通信网TES系统为节省卫星转发器频率资源对传输的语音信号进行压缩处理,其信道单元基带信号处理器对语音信号进行CCITT推荐的G.721-ADPCM编码和修斯公司专利技术开发的RLPC编码处理,将64 kb/s语音数字信号压缩至32 kb/s,16 kb/s,9.6 kb/s传输,实现语音质量满足一般通信要求的低速率语音信号传输。     

一种新的8~32kb/s宽带嵌入式变速率语音编码方法

 本文基于ACELP和TCX编码技术,提出了一种8~32kb/s五层宽带嵌入式变速率语音编码方法,其中,前三层采用ACELP实现了8kb/s、12kb/s和16 kb/s的嵌入式编码,后两层采用TCX技术实现了24 kb/s和32 kb/s嵌入式编码.实验结果表明,该嵌入式语音编码方法的质量在纯净语音、办公室噪声和层间转换方面接近于ITU-T G.VBR的TOR要求.     

ML7204和PicoBlaze软核处理器实现低速话音编解码系统

语音信号压缩编码是数字语音信号处理的主要方面．在现有的语音编码中,G．729A算法在8kb／s的码率下取得了较好的语音质量,具有广阔应用前景,因此提出采用PicoBlaze和ML7204实现G．729A语音压缩／解压详细的软硬件实现方案,并描述了G．729A语音编解码器ML7204的工作原理、性能、接口,以及FPGA内嵌IP核微处理器PicoBlaze的特点和使用方法。给出硬件电路设计原理,以及各部分的具体实现方法和原理图。并给出软件流程和主要代码。实验结果表明,系统提供话音点到点的时延仅为25mS,而语音质量平均意见MOS值达到4．2。在可懂度和清晰度等性能优异,该系统设计可应用于无线移动网、数字多路复用系统和计算机通信系统。     

多媒体终端中声音和数据的集成传输

本文描述了采用包复用方式在固定带宽内集成传输声音和多媒体数据的多媒体终端通信系统，系统中的声音编码采用了静默检测技术，声音编码的速率可以根据信道的拥挤程度在３２ｋｂｉｔ／ｓ和１６ｋｂｉｔ／ｓ之间动态地变化。本文提出了一种利用增减静默抽样来同步声音编解码时钟的方法，本文还提出了利用数据队列的短时平均长度来判断信道繁忙程度的算法，在多媒体数据突发性强、数据量大时，该算法比利用声音或数据队列的瞬时长度判断更为准确。     

Integration of Voice and Data in the Wideband Packet Satellite Network

The Wideband (packet satellite) network is an experimental 3 Mbit/s communications system developed under sponsorship of the Defense Advanced Research Projects Agency and the Defense Communications Agency. This system is being used to evaluate the use of packet transmission for efficient voice communication, voice conferencing, and integration of voice and data over a satellite channel. Each station in the Wideband network consists of an earth terminal (dedicated 5 m antenna plus associated IF/RF equipment), a burst-modem and codec unit, and a station controller. Station controllers provide interfaces to host computers (including packet speech sources) and manage the allocation of the satellite channel on a TDMA demand-assigned basis. TDMA demand-assignment is implemented using a reservation-based packet-oriented protocol capableof handling traffic at multiple priority levels. The channel protocol provides a reservation-per-message mode of service (datagrams) to support transmission from bursty traffic sources and a reservation-per-call mode of service (streams) to support traffic with more regular arrival statisticS (e.g., vioce). A distributed scheduler running in every station controller eliminates the need for a central control station and minimizes network transit delay for datagram transmission as well as stream creation, modification, and deletion. In this paper we describe the protocols and mechanisms upon which the Wideband packet satellite network is based.     

Lattice low-delay vector excitation coding of speech at 8-16 kb/s

Lattice low-delay vector excitation coding (LLD-VXC) is a speech coding system based on analysis-by-synthesis excitation coding and backward adaptation of the synthesis filter. The introduction of a lattice filter as a (high order) short-term predictor has significant advantages, such as fast tracking of speech signal nonstationarities, simple stability verification, and uniform distribution of the computational load. The objective of this paper is to present a Lattice LD-VXC (LLD-VXC) codec-and experimental results obtained at rates of 8, 9.6, and 16 kb/s. A sign algorithm for lattice filter adaptation is introduced in order to reduce computational complexity. An LLD-VXC codec with a 20th-order lattice predictor, a 10th-order lattice weighting filter, and a backward pitch predictor achieved toll quality at 16 kb/s and good communications quality at 8-9.6 kb/s with a delay of less than 2 ms and reasonable complexity     

一种符合ITU-T指标要求的嵌入式立体声语音频编码方法

 基于国际电信联盟标准化组织(ITU-T)编码标准G.729.1,本文提出了一种嵌入式变速率立体声语音与音频编码方法.本算法利用G.729.1和改进的调制叠接变换(Modulated Lapped Transform,MLT)编码技术对输入信号的中值与边带信息进行分层编码,形成具有嵌入式结构的码流.编码器可处理宽带和超宽带的立体声信号,宽带立体声信号编码的最大码率为48kb/s,超宽带立体声信号编码的最大速率为64kb/s.实现结果表明,本编码器的编码质量均达到了ITU-T对G.EV-VBR立体声编码的指标要求.     

Entropy coding of ITU G.728 speech parameters for low bit rateaudiovisual communications

Entropy coding principles are applied to the 16 kbit/s ITU G.728 speech codec. It is shown that the average bit rate can be reduced to 14.5 kbit/s without a significant increase in the codec complexity. In very low bit rate audiovisual communication applications such as the videophone, the saved bits can be used to improve the output video quality     

1200/2400bps改进型多带激励声码器的实时实现

本文介绍了基于多带激励（ＭＢＥ）语音模型的改进型全双工１２００／２４００ｂｐｓ声码器．该声码器已应用于多种通信系统中．其语音清晰度（ＤＲＴ标准）１２００ｂｐｓ时为９１７５,２４００ｂｐｓ时为９２６７．本文重点介绍其硬件结构及算法实现．     

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.     

Pulsed residual excited linear prediction

Linear predictive coding of speech has been widely used at 16 kb/s in the form of adaptive predictive coding (APC) down to 4.8 kb/s in the form of code-excited linear prediction (CELP). Since its invention in 1984 there have been many variations of CELP which differ mainly in the way the final excitation signal (codebook) is produced and quantised. These variations either produce better speech quality or lower complexity. Three new excitation types, all of which are based on a pulsed residual, are proposed. The new pulsed residual excitations improve the speech quality significantly. In addition a novel mathematically equivalent codebook search method which reduces the search complexity significantly is described     

The DTX-240D digital circuit multiplication system

This paper describes the DTX-240D digital circuit multiplication system (DCMS) offered by ECI Telecom. It will accept up to 240 × 64 kb/s trunks carrying either 64 kb/s voice, voice band analogue non-speech signals, or digital data for transmission over a 2·048 Mb/s digital link. Over 1000 are currently ‘on-line’ and carrying traffic. The system comprises a pair of terminals, one on each side of the interterminal digital link (bearer). It will normally operate in the network at a concentration ratio of 5:1, in which case 150 × 64 kb/s trunks, carrying voice, voice band data or digital data can be concentrated into one 2·048 Mb/s bearer. The users are able to increase the number of trunks up to 240 per 2·048 Mb/s bearer, when time zone differences cause a spread of busy-hour traffic carried on a single system. Each terminal will normally be located at an international switching centre (ISC) but may also be located at an earth-station. The system uses a DSI (digital speech interpolation) stage providing a 2·5:1 multiplication, followed by an additional 2:1 multiplication by means of ADPCM (adaptive differential pulse code modulation). In addition, the VBR (variable bit rate) technique is used to prevent clipping, due to overload congestion. The system can also be used with 1·544 Mb/s digital bit streams (trunk side or bearer).     

TRANSMISSION PERFORMANCE EVALUATION OF VOICE ENCODING TECHNOLOGY FOR THE INMARSAT MINI-M SYSTEM

In this paper are presented the method and results of a subjective evaluation, which was conducted in order to select a new speech codec for the Inmarsat mini-M system. The mini-M system is designed to provide the next generation of global, notebook-sized satellite terminals for transportable, land-mobile and maritime voice, facsimile, and data communications. Overall, six different codecs operating at a combined source and channel rate of 4⋅8 kbit/s were evaluated in a series of six subjective tests. From this, it was concluded that one codec was able to deliver performance that is equivalent to, or better than, the IS-54 full-rate ditigal cellular 8 kbit/s VSELP codec, and was selected for use in the mini-M system.     

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APD的紫外光语音通信系统设计

设计了一套基于雪崩光电二极管(APD)的紫外光语音通信系统,系统分为硬件电路、软件和光学系统三个部分.采用先进多带激励(AMBE)编码和开关键控(OOK)调制技术,使语音通信速率降低到2.5kb/s.实验表明,该系统在光功率仅为纳瓦量级时仍能完成实时语音通信,系统的研制对日盲APD的制造和应用具有现实的指导意义.