Pitch synchronous 1st-order linear d.p.c.m. system

A system called p.s.f.o.l.d. is described which exploits the correlation between successive pitch periods of a speech signal. This system is a differential one and can employ various types of encoders. We describe a p.s.f.o.l.d. system using a 1st-order d.p.c.m. encoder and show that for a speech utterance this system has a peak signal/noise ratio which is 6 dB larger, and has an increase in dynamic range of 13 dB, compared with a 1st-order d.p.c.m. codec.     

Noninteger sampling to subcarrier frequency ratios in d.p.c.m. of PAL signals

A simple predictor having three coefficients is described for d.p.c.m. encoding of composite PAL colour signals. Noninteger ratios of sampling frequency fs to subcarrier frequency fsc can be selected. For a colour-bar signal, it is shown that the s.n.r. of the d.p.c.m. system using fs/fsc = 2.5 is approximately the same as when third-previous sample prediction is used, i.e. when fs/fsc = 3.0.     

Estimating the pitch period of voiced speech

In speech processing an estimation of the speech pitch period is important. Real time pitch detection is only possible by the selection of an efficient algorithm suitable for implementation on a programmable processor or in special-purpose hardware. The use of the periodogram algorithm (p.a.) is proposed to detect the pitch period of voiced speech. This algorithm is attractive for the following reasons: (a) it has no multiply operation; (b) when implemented on a 16-bit computer (e.g. microprocessor) the computation can be done in integer arithmetic without exceeding the microprocessor's dynamic range; (c) it is a simple technique for estimating the pitch period with reasonable accuracy. Results of the analysis of speech signals and sinusoids using the periodogram algorithm are presented and comparisons are made with the average magnitude difference function (a.m.d.f.) which is an alternative method of estimating the pitch period of the voiced speech.     

基于虚拟仪器技术的双通道语音分析系统的设计

针对语音信号的特点,采用主流虚拟仪器开发软件——LabVIEw8.0开发出双通道语音分析系统,该系统既可满足对动态语音信号进行高精度、高频率采样的要求,也可对每个通道的动态输入范围进行单独设置,以保证量化噪声对动态范围不同的信号的影响尽可能保持一致。本系统采用ActiveX技术实现了与数值分析软件matlab6.5的连接,利用matlab中小波工具包对采集的语音信号的进行噪声滤除等处理,整个系统人机界面友好、结果直观、并且通过改进程序可以扩展系统的功能。     

Pseudorandom 250 Mb/s modulation of GaAs l.e.d.s

Effective direct modulation of a GaAs l.e.d. was demonstrated with a 250 Mb/s optical-repeater experiment. A GaAs l.e.d. transmitted a high-quality p.c.m. pseudorandom bit stream, which permitted the received signal to be regenerated with predicted error rates.     

Adaptive difference detection and correction system for partial correction of transmission errors in linear p.c.m.

An adaptive version of the difference detection and correction system for the partial removal of transmission errors in linear p.c.m. speech without the use of channel coding is presented. The improvement in s.n.r. compared to the nonadaptive system is approximately 3 dB for transmission bit error-rates between 0.5 and 5%.     

Detection and partial correction of isolated errors from the received data in a 1st-order d.p.c.m. codec

A detection and partial-correction (d.p.c.) system located at the receiver in a 1st-order d.p.c.m. system enables isolated transmission errors that would cause gross distortion in the recovered signal of a conventional d.p.c.m. decoder to be corrected from the received data, providing that the input signal does not exhibit a rapid change in its slope during a sampling period. The d.p.c. system is disabled when this occurs.     

Performance degradation of two-link binary c.p.s.k. system incorporating direct phase regenerator

The letter presents an analysis for the bit error probability (b.e.p.) of a two-link binary coherent phase-shift-keying (c.p.s.k.) system corrupted by additive Gaussian noise and consisting of a direct phase regenerator (d.p.r.) followed by a coherent receiver. Equivalent signal/noise ratio (s.n.r.) degradations due to imperfections in a practical d.p.r. are evaluated at the 10?6 b.e.p. point.     

IPv6的安全体系结构

文章对IPv4协议安全方面的一些缺点进行了分析,讨论了OSI的网络安全体系结构,给出IPv6新的网络安全机制,详细描述了IPSec所提供的网络安全服务与实现原理,并对IPSec的两个安全协议———AH和ESP作了较深入的阐述。     

High-performance c.c.d. channel filter for a p.c.m. codec

A monolithic c.c.d. filter for p.c.m. codec was fabricated employing both a minimum-phase design with a substantially reduced number of taps and almost perfect elimination of excess capacitances. Results, such as negligible degradation in frequency responses, considerable reduction of common-mode signal, low filter noise (?78 dBm) and wide dynamic range (84 dB) were successfully obtained.     

基于AD8368芯片的压控增益放大系统设计分析

针对接收机中信号电平变化范围过大所导致的系统恶化问题,提出基于中频信号检测电压反馈,2级AD8368芯片级联构造具有70 dB动态范围的中频大动态自动增益控制(AGC)的方法。对自闭环控制电路进行了分析,证明该方法能够解决宽动态范围下信号增益自动控制问题,实现了快速、稳定的自动增益控制,从而保证接收机小信号下的高灵敏度,提高系统的接收灵敏度。     

Adaptive nonuniform polar quantization application to high quality speech compression

The logarithmic companding technique has shown to be extremely useful in speech quantization with rate of 8 bits/sample. However, for lower bit rates it is not the ideal solution for high quality speech coding. Because of that, in this paper we establish source coding scheme which enables better spectrum efficiency for input that has a large dynamic range. Since our wish is also to improve signal quality in comparison with quality defined with standards G.711 and G.712, we opt for adaptive technique application to the speech coding. Our research shows that proper design of forward gain-adaptive polar quantization can enable compression of about 1 bit/sample as well as significantly better quality than in case of using coder designed according to standard G.711. Furthermore, performances can be sustained over the whole speech dynamic range. Also, if the requisite speech quality is not supposed to be lower than G.712 standard quality, the achieved compression can be almost 1.5 bits/sample. Besides, we propose knew simple encoding rule which can additionally reduce bit rate for 0.1 bit/sample.     

Reducing quantising noise in p.c.m. coders

An analogue-feedback method has been developed to reduce quantising noise in p.c.m. systems. The improvement in signal/noise ratio, however, depends on the loop delay, saturation limit of the coder and the number of digits used. The overall characteristics of the feedback p.c.m. systems have been found to be better than those of the conventional p.c.m. for bit rates up to 50 × 103 bit/s.     

Speech-transmission performance of p.c.m. systems

A method is described for calculating a quantity representative of the transmission performance of a p.c.m. system. The value obtained depends upon (a) the number of quantised output voltage states, (b) the companding law and (c) the volume of the input speech signal relative to the overload point of the system.     

Analogue c.c.d. correlator using monolithic m.o.s.t. multipliers

A 32-stage analogue correlator has been hybridised by using a multitapped c.c.d. delay line and m.o.s.t. multipliers fabricated on the same silicon chip. A multiple-port sample?hold system has been adopted for storage of the reference signal, thereby allowing it to be refreshed continuously. The viability of a design for a fully analogue monolithic c.c.d. correlator is established.     

Compact helium speech unscrambler using charge-transfer devices

Helium speech is the term commonly used for the distorted speech uttered by deep-sea divers breathing in a helium/oxygen mixture. Present unscrambler designs use pitch synchronous time-expansion signal processing with digital storage. The compact unscrambler reported here has been configured using analogue charge-transfer devices for waveform storage and c.m.o.s. digital circuitry for control logic as a precursor to development of the whole system as a single integrated circuit. The compact unscrambler itself is shown to offer distinct engineering and operational advantages.     

从搜索引擎看IPv6网络增长

文章根据“网络指南针”IPv6搜索引擎近一年来积累的IPv6网页数据,概括了全球IPv6站点数量的增长情况,分析了IPv6站点与IPv4站点引用率的差别,最后得出关于IPv6网络增长的一些结论。     

加性噪声条件下鲁棒说话人确认

基于非负矩阵分解的语音去噪,在提高语音信号信噪比的同时,也会引起语音失真,从而导致噪声环境下说话人确认系统性能下降.本文提出基于分区约束非负矩阵分解的语音去噪方法（Nonnegative Matrix Factorization with Partial Constrains,PCNMF）,目的是在未知和非平稳噪声条件下提高话人确认系统的鲁棒性.PCNMF在满足分区约束条件的基础上分别构建语音字典和噪声字典.考虑到传统语音训练产生的语音字典往往含有一定的噪声成分,PCNMF通过数学模型产生基音及泛音频谱,在此基础上利用该频谱模仿人声的共振峰结构来合成字典,从而保证语音字典纯净性.另一方面,为了克服传统噪声字典构建方法带来的部分噪声信息丢失问题,PCNMF对在线分离出的噪声样本进行分帧和短时傅里叶变换,然后以帧为单位线性组合生成噪声字典.性能评估实验引入了多种噪声类型,实验结果表明PCNMF可有效提高说话人确认系统的鲁棒性,特别是在未知和非平稳噪声条件下其等错率相比基线系统（Multi-Condition）平均降低了5.2%.     

Upper bound on signal/quantising-noise ratio for television d.p.c.m. systems

Consideration is given to an upper bound on signal/quantising-noise ratio for television d.p.c.m. systems. The 7.2 dB gain for entropy coding assumes ? entropy of the bit stream of the quantiser output. The calculations are based on Laplacian signals, because television signals at d.p.c.m. quantiser inputs are approximately Laplacian.     

Optimised hybrid unidigit p.c.m. system

The coding efficiency of unidigit p.c.m. systems has been improved by using a secondary feedback loop, and the gain/frequency characteristic of the coder has been equalised. The overall signal/noise characteristic of this hybrid system is found to be better than those of all other unidigit systems; it is inferior to the conventional p.c.m. system only for bit rates higher than 60 × 103 bit/s.