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).     

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.     

New trends in digital circuit multiplication technology

This paper discusses the achievements and the unsolved problems of the first generation of Digital Circuit Multiplication Systems (DCMS)

1 DCMS is an abbreviation for Digital Circuit Multiplication Systems. DCME is an abbreviation for Digital Channel Multiplication Equipment. Both refer to the same technique of combining digital speech interpolation and low rate encoding.

. The following summarizes proposed improvements in DCMS technology. A major improvement in DCMS gain may be achieved by implementation of facsimile demodulation. The design considerations of this technique are discussed. The CCITT activities on standardization of a 16 kb/s speech coding algorithm are reviewed. In order to adapt this algorithm to implementation in a DCMS, the operation of the algorithm has been modified to enable variable bit rate operation at rates between 12·8 and 24 kb/s. The implementation of such an algorithm in DCMS will further increase the overall compression gain. The first generation of DCMS has been implemented mainly on heavy traffic routes. A DCMS terminal which is optimized for operation on ‘thin’, or smaller, routes, using an IDR link operating at 512 kb/s, is described. Tandem operation of DCMS is becoming an increasingly important consideration on both international and national routes. The accumulation of qdu's (quantization distortion units) when two pairs of DCME/S terminals are operated in tandem can be avoided by an arrangement without ADPCM decoding/encoding in the transit terminals so that the ADPCM decoding is carried out only at the final destination DCMS terminal.     

16 Kb/s high quality voice encoding for satellite communication networks

This paper describes applications of adaptive predictive coding (APC) with maximum likelihood quantization (MLQ) which can cover a wide range of coding rates from 4.8 to 16 kb/s for low C/N satellite communication systems, such as maritime, aeronautical mobile and thin-route satellite communication systems, and also for speech and data integration, including digital circuit multiplication equipment (DCME) in business communication systems, such as INTELSAT business services (IBS). A 16 kb/s APC–MLQ hardware codec has been implemented by NEC–7720 DSP chips and the performance has been confirmed in subjective quality of speech through conversational tests. The objective performance has also been evaluated for non-voice signals, such as single and multi-frequency tones, and 1200 and 2400 b/s voiceband data signals. The APC-MLQ codec can transmit the voice-band data at 1200 b/s over two asynchronous tandem links and at 2400 b/s over one link. It was noted that the APC-MLQ codec is superior in speech performance at 16 kb/s to a narrow-band companded FM and meets requirements for low C/N satellite communication systems. For voice and data integration into 16 kb/s for 64 kb/s links, we propose a multi-media multiplexing for low C/N digital satellite communication systems and also a small-scale circuit multiplication system for business use. In these systems, a variable rate coding of APC-MLQ from 4.8 to 16 kb/s can be effectively introduced for voice and data integration.     

Out-of-Band Signaling Scheme for High Speed Wireless LANs

In recent years, the physical layer data rate provided by 802.11 Wireless LANs has dramatically increased thanks to significant advances in the modulation and coding techniques employed. However, previous studies show that the 802.11 MAC operation, namely the distributed coordination function (DCF), represents a limiting factor: the throughput efficiency drops as the channel bit rate increases, and a throughput upper limit does indeed exist when the channel bit rate goes to infinite high. These findings indicate that the performance of the DCF protocol will not be efficiently improved by merely increasing the channel bit rate. This paper shows that the DCF performance may significantly benefit from the adoption of two separate physical carriers: one devised to manage the channel access contention, and another devised to deliver information data. We propose a scheme, referred to as out-of-band signaling (OBS), designed to reuse (and remain backward compatible with) the existing 802.11 medium access control (MAC) specification. Performance evaluation of OBS is carried out through analytical techniques validated via extensive simulation, for both saturation and statistical traffic conditions. Numerical results show that OBS improves the throughput/delay performance, and provides better bandwidth usage compared with the in-band signaling technique employed by DCF.     

High-Gain Digital Speech Interpolation with Adaptive Differential PCM Encoding

This paper describes a digital speech interpolationadaptive differential PCM bit reduction technique in which digital speech interpolation (DSI) is combined with ADPCM encoding. A highly sensitive speech detector, a voiceband data discriminator, and a variable rate ADPCM encoding are used to achieve a high compression ratio. The speech detector proposed in [1] detects speech signals above -51 dBm with 32 ms hangover time; average talk spurt activity of 36 percent was measured on fully loaded trunks in an international satellite link. Features of the speech power spectrum are used for adaptively controlling the bit length from 2 to 4 in an ADPCM speech encoder. Voiceband data are detected with 10 ms by the voiceband data discriminator. 5 bit ADPCM encoding is applied to voiceband data to maintain transparency through the DSI-ADPCM system. A DSI gain of 3 is expected as a result of the highly sensitive speech detection, the variable rate encoding technique, and the voiceband data discrimination. Speech and voiceband data are efficiently transmitted through an ADPCM encoding with either a 6 or 6.4 kHz sampling rate converted from an 8 kHz sampling rate. To avoid a band limitation as much as possible, a frequency shift manipulation on the voiceband channel is incorporated prior to the sampling conversion. Consequently, a total bit reduction gain of 7 to 4 is expected relative to a 64 kbit/s PCM transmission. Satisfactorily high quality of the processed speech has been obtained through computer simulations.     

Some aspects of time-division data switch design

A number of techniques for configuring circuit switching data network carrying mixed bearer rate traffic are discussed. In the network, blocking probability unbalance among different bearer rates arises on highway and switch, for example, due to fractional occupation of usable channel for high bearer rate call by lower bearer rate call. Such a problem due to handling mixed bearer rate traffic is first described. Several solutions to this problem are presented and evaluated in terms of memory amount, delay time through switch, and dynamic program steps required for path search procedure. This paper concludes that single-stage time-division switch (T switch) is suitable for data purpose, so far as switch size is relatively small. Also presented are fundamental techniques for designing a T switch with reduced memory chips and reduced memory speed requirement. T switch configuration varieties based on these techniques are quantitatively evaluated. A design method for the most economical and flexible T switch under conditions such as usable memory, traffic distribution in data speed, and local network configuration, is established.     

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     

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.     

一种线性化的全双工MIMO收发器设计

针对全双工MIMO收发器发射通道非线性以及接收通道存在强烈自干扰的问题,该文提出一种使发射通道线性化并通过射频多抽头重建与数字重建消除自干扰的具有较低硬件成本与软件复杂度的设计方案:(1)基于改进的串扰消除和数字预失真(CTC-DPD)算法并复用反馈通道进行去耦合和数字预失真使发射通道线性化、等增益;(2)在接收通道加入可调衰减器并用多维梯度下降法基于接收的残留自干扰功率最小原则调整抽头参数;(3)基于频域信道估计进行数字自干扰重建。实现的20 MHz带宽LTE全双工22 MIMO通信样机,发射通道经过线性化后带内更平坦,而带外噪声抑制了约30 dB。射频和数字消除一轮调整共耗时约0.17 ms,总消除能力约75 dB。16QAM映射时全双工双向数据速率总和220 Mbps,相对单向时的110 Mbps实现了频谱效率的翻倍。通信样机证明了该方案的可行性。     

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     

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.     

Expérimentation sur site ďun système de multiplication de circuits par mise en paquets

This article describes the main results of the field trial conducted by France Telecom andatt of theiacs. This packetized circuit multiplication equipment has been submitted to the following tests: subjective evaluation of speech (subscribers interview), performances of voice band data transmission, performances with a bit error rate on the bearer, test of the fax demodulation process.     

Huffman code based error screening and channel code optimization for error concealment in perceptual audio coding (PAC) algorithms

The class of perceptual audio coding (PAC) algorithms yields efficient and high-quality stereo digital audio bitstreams at bit rates from 16 kb/sec to 128 kb/sec (and higher). To avoid "pops and clicks" in the decoded audio signals, channel error detection combined with source error concealment, or source error mitigation, techniques are preferred to pure channel error correction. One method of channel error detection is to use a high-rate block code, for example, a cyclic redundancy check (CRC) code. Several joint source-channel coding issues arise in this framework because PAC contains a fixed-to-variable source coding component in the form of Huffman codes, so that the output audio packets are of varying length. We explore two such issues. First, we develop methods for screening for undetected channel errors in the audio decoder by looking for inconsistencies between the number of bits decoded by the Huffman decoder and the number of bits in the packet as specified by control information in the bitstream. We evaluate this scheme by means of simulations of Bernoulli sources and real audio data encoded by PAC. Considerable reduction in undetected errors is obtained. Second, we consider several configurations for the channel error detection codes, in particular CRC codes. The preferred set of formats employs variable-block length, variable-rate outer codes matched to the individual audio packets, with one or more codewords used per audio packet. To maintain a constant bit rate into the channel, PAC and CRC encoding must be performed jointly, e.g., by incorporating the CRC into the bit allocation loop in the audio coder.     

Deriving a subjective testing methodology for digital circuitmultiplication and packetized voice systems

The CCITT expert group on speech quality, formed in part to derive methodologies for evaluating new speech technologies, is emphasizing the assessment of digital circuit multiplication and packetized voice systems. Study group XII was requested to assess the performance of digital circuit multiplication equipment (DCME). The authors discuss DCME terminology, applications, testing alternatives, and issues that need to be addressed     

Interworking IEEE 802/FDDI LANs via the ISDN frame relay bearerservice

The author presents a generic architecture for interconnecting LANs (local area networks) through the ISDN (integrated services digital network) bearer services, particularly the frame relay bearer service. The architecture is derived from the IEEE 802.1 MAC (medium access control) bridge and ISDN frame relay standards. An algorithm for MAC/ISDN address resolution that makes minimal use of the WAN (wide area network) bandwidth (which is potentially the most expensive resource) is presented. The algorithm uses a MAC/ISDN address resolution server to resolve addresses for new stations, the resolution in all other cases (e.g. stations moving from the ISDN address to another) being fully distributed. To prevent a server failure from inhibiting communication to new stations, a backup server may be provided. A practical implementation of the architecture has been found capable of supporting full throughput at ISDN hyperchannel rates (384-1920 kb/s) for all IEEE 802.3 frame lengths. Frame relay is seen as having a number of important advantages for LAN interconnection, including the following: a large number of virtual circuits available, giving the potential for a rich interconnection architecture with single-hop connections across the ISDN; and low processing overhead enabling efficient use of ISDN channels, including ISDN hyperchannels (384-1920 kb/s)     

A comparison of the implementation and annual operating costs of IDR and TDMA in the intelsat system

The outcome of a preliminary systems cost study which compares the cost of operating INTELSAT intermediate data rate (IDR) and time division multiple access (TDMA) stations under various conditions is reported. The differences in the annual cost of operating the two candidate methods over the range of 1 to 1000 terrestrial channels are derived under the conditions of no CME, DSI only and the use of DCME. For DCME the effect of one, two, three and four destinations per DCME is considered. The results show that while the initial cost of the TDMA equipment is higher than with IDR, the annual cost of operating the systems is dominated by the space segment costs. For this reason the greater bandwidth efficiency, inherent multidestinational capability and ease of capacity expansion of the TDMA system give it a lower annual operating cost, even at fairly moderate earthstation capacities (e.g. for DCME with two destinations TDMA becomes cheaper than IDR for capacities in excess of about 300 channels). There are attributes of both systems which will also influence the network planners' decision. The IDR system is modular in that costs associated with implementation are incurred more gradually than with the TDMA system. Furthermore, IDR is to a great extent an extrapolation of the current FDM/FM/FDMA practice. However, since the equipment costs are small compared to the space segment charges, it may prove advantageous to adopt the lowercost system as soon as possible. In addition, the TDMA system offers significant long-term advantages of low-cost expansion and the ability to reconfigure the system with minimal or no loss of traffic.     

An ADPCM Receiver for NTSC Composite Video Signals

An ADPCM codec for carrying one broadcast quality NTSC color TV channel at a bit rate of 42.9 Mb/s has been proposed. The system uses 3 fsc sampling, adaptive intrafield contour prediction, adaptive quantization., 4/8-bit dual length coding, and horizontal blanking interval suppression techniques. The receiver of.the video codec is designed and implemented in ECL for recovery of the original signal. The receiver accepts.a 42.9 Mb/s serial data stream with a synchronous clock from the transmitter. The receiver detects the line synchronization code, demultiplexes the audio signal and video signal, and generates the horizontal blanking patterns which have been removed at the transmitter side. The 4/8-bit dual length code is decoded and fed to the ADPCM reconstruction loop to obtain the reconstructed active video signal. The generated horizontal blanking pattern is multiplexed with the reconstructed video=signal and sent through a D/A converter to form the reconstructed analog NTSC composite video signal.