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.     

The use of digital circuit multiplication equipment in satellite communications

The use, within satellite communications, of low rate encoding (LRE) techniques, based on 24, 32 and 40 kb/s ADPCM coding, coupled with digital speech interpolation (DSI) to form a digital circuit multiplication equipment (DCME), is addressed in this paper. The need for a system simulation tool, in order to plan for and correctly use the DCME concept is identified. Results obtained with this simulation tool are presented. The simulation model makes it possible to predict the behaviour of the system from a quality point of view, with external conditions simulated to be very close to actual operating conditions.     

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).     

Two approaches for waveform coding of 9.6 kbit/s voiceband datasignals through 32 kbit/s ADPCM

Two approaches are presented for accommodating 9.6 kb/s modem signals (e.g. V.29) through 32 kb/s ADPCM (adaptive digital pulse-code modulation) links. These are small changes in the existing algorithm and coding with ADPCM incorporating a 5-bit, rather than 4-bit quantizer. For each approach, tradeoffs between performance and implementation complexity are described     

ADPCM with a multiquantizer for speech coding

A speech coding algorithm with low complexity and a short processing delay is introduced. The proposed algorithm is ADPCM (adaptive digital pulse code modulation) with a multiquantizer (ADPCM-MQ). The input signal is processed in parallel by multiple ADPCM coders with different characteristics. Then the optimum ADPCM coder with minimum error power is dynamically selected for each frame. A 16-kb/s codec based on this algorithm has been implemented using two general-purpose digital signal processors (MB8764) with 8.3 ms of total processing delay. A segmental SNR of 19-21 dB was achieved at 16 kb/s; with postfiltering the segmental SNR was increased to 23-25 dB. Combined with the time domain compression scheme, the algorithm can be easily applied to 8-kb/s coding. It is also extensible to variable-rate coding     

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     

ADPCM语音解码合成输出系统的设计

文章介绍了自适应差分脉冲编码调制(ADPCM)技术的编解码和脉冲宽度调制(PWM)技术的基本原理,研究在现场可编程门阵列(FPGA)上通过有限状态机方式实现ADPCM语音解码算法,利用PWM技术将解码后的数字语音信号转化为PWM波,以此直接驱动喇叭发出声音,输出的合成语音质量良好.     

Enhancement of ADPCM speech coding with backward-adaptivealgorithms for postfiltering and noise feedback

It is shown that postfiltering circuits based on higher order LPC (linear predictive coding) models can provide very low distortion in terms of special tilt. Thus, they can provide better speech enhancement than circuits based on the backward-adaptive pole-zero predictor in ADPCM (adaptive digital pulse code modulation). Quantitative criteria for designing postfiltering circuits based on higher-order LPC models are discussed. These postfilters are particularly attractive for systems where high-order LPC analysis is an integral part of the coding algorithm. In a subjective test that used a computer-simulated version of these circuits, enhanced ADPCM obtained a mean opinion score of 3.6 at 16 kb/s     

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.     

The Residual Encoder--An Improved ADPCM System for Speech Digitization

An improved system for speech digitization using adaptive differential pulse-code modulation (ADPCM) is described. The system uses an adaptive predictor, an adaptive quantizer, and a variable length source coding scheme to achieve a 4-5 dB increase in signal-to-noise ratio over previous ADPCM. The increase can be used to improve speech quality at moderate data rates on the order of 16 kbits/s or to retain the same quality and reduce the data rate to 9.6 kbits/s. The latter alternative permits the use of narrow-band channels. The implementation complexity is on the same order as other ADPCM systems.     

Transmission quality of north American cellular, personalcommunications, and public switched telephone networks

Future long distance, and especially international calls, will involve an increasing number of multilink circuits of cellular, personal communications, mobile satellite, and public switched telephone network (PSTN) type of connections incorporating a variety of speech coding devices. In particular, the rapid growth of cellular communications has highlighted the need to characterize the quality of switched networks when cellular terminals are attached at their termination nodes. At the same time, the nonlinear nature of low-rate parametric speech coding has rendered questionable analytical methods for estimating end-to-end voice quality of interconnected telecommunications networks. Instead, quantification of transmission performance appears to require direct subjective evaluation of the pertinent conditions of interest. In this paper the quality of interconnected North American digital cellular and future microcellular terminals with 16 kbit/s and 32 kbit/s DCME/PCME-based switched and private telephone networks is quantified. From these assessments it can be concluded that cellular networks employing the TIA IS-54 8 kbits/s VSELP algorithm may meet the end-to-end transmission planning criteria when interconnected with the switched network     

High-quality coding of telephone speech and wideband audio

Digital speech technology is reviewed, with the emphasis on applications demanding high-quality reproduction of the speech signal. Examples of such applications are network telephony, ISDN terminals for audio teleconferencing, and systems for the storage of audio signals, which include the important subclass of wideband speech. Depending on the application, the bandwidth of input speech can vary from about 3 kHz to nearly 20 kHz. Coding for digital telephony at 4 and 8 kb/s, network quality coding at 16 kb/s, and coding for audio at 7 and 20 kHz are examined. Future directions in the field are discussed with respect to anticipated technology applications and the algorithms needed to support these technologies     

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。     

Design and simulation of an efficient adaptive delta modulationembedded coder

An embedded coding version of hybrid companding delta modulation (HCDM) is described that operates from 16 to 48 kb/s in 8 kb/s steps. The embedded HCDM coder employs the explicit noise coding technique to transmit an adaptive PCM (APCM) coded version of the HCDM reconstruction error signal as a supplementary bit stream that may be partly or wholly deleted in transmission. SNR performance with speech input depends critically on the design of the supplemental APCM code and two new coding algorithms are investigated. In algorithm 1, the basic cue for step size adaptation is obtained from the RMS slope energy of the HCDM output whereas in algorithm 2, the HCDM reconstruction error is logarithmically compressed before quantisation and the basic step size is derived from peak input magnitudes. Instantaneous adaptation for both algorithms is achieved by using step size multipliers which are optimised for operation at single fixed bit rates and also for decoding with an unknown number of input bit deletions. Simulation results show that SNR performance is significantly enhanced using either algorithm and a graceful reduction of reconstructed speech quality with progressive bit deletion is achieved over the range from 48 kb/s to 16 kb/s. On the whole, the SNR performance of the embedded HCDM system is superior in comparison with conventional HCDM     

实现Reed-Solomon码译码的新电路--在普通基上用"比特串行乘法电路"实现RS码译码

本文推出了在域GF(2m)上用于RS码译码的两种新电路:普通基"比特串行序列乘法电路"和"比特串行乘法累加电路",基本上以m个与门代替了两个任意元素相乘的复杂乘法器,使译码电路大大简化.作为一个应用实例,详细阐明了用它们构造的RS码纠删/纠错译码各步电路.这两种新电路对性能优良的RS码的使用和推广具有实用价值.     

基于改进RBF神经网络的ADPCM语音编码算法研究

以G．721ADPCM语音编码算法为研究对象，在语音编码的预测中引入神经网络模型来克服传统线性滤波方法中存在的不足，研究了基于RBF神经网络的ADPCM语音编码系统的结构。通过k均值聚类算法来确定RBF神经网络的中心和宽度，用最小二乘法确定RBF网络权值的方法改进了ADPCM语音编码算法。实验证明．其平均信噪比较原ADPCM编码算法有1-2dB的提高。     

The Design of an ADPCM/TASI System for PCM Speech Compression

This paper describes the design of a digital speech interpolation (DSI) system called ADPCM/TASI for adaptive differential PCM with time assignment speech interpolation. This system is designed to compress the output of two T1 24-channel PCM carrier terminals into a 1.544 Mbit/s signal that can be transmitted over a single T1 carrier line. The design is based on a bit slice microprocessor structure. Alternative designs are also described.