A Comparison of Optimum and Logarithmic Quantization for Speech PCM and DPCM Systems

Two quantization rules, optimum (minimum mean-square error) and logarithmic companding, are compared for application to digital speech transmission. Comparison is made on the basis of signal-to-quantizing noise ratio (S/N), subjective quality judgments, idle channel noise, and dynamic range. These quantizers are considered in both PCM and differential PCM configurations. A computer algorithm is described that yields the optimum quantizer levels for a given speech record. It has been found that the improvement in S/N which optimum quantization affords over conventional logarithmic quantization is offset by the greater idle channel noise and smaller dynamic range (range of talker volumes handled with a lower limit on S/N) of the optimum law.     

Optimum PCM Codes for DPSK

The effect of digital errors in PCM systems for speech signals depends on the PCM code used. The standard binary folded PCM code is superior to the natural binary code but it is not optimum. This paper considers PCM codes especially designed to be insensitive to the double error patterns produced by DPSK. It is concluded that the dynamic range with especially designed PCM codes is considerably extended compared to standard binary folded PCM. Special simplified interleaving schemes and optimum PCM codes for coherent PSK are also considered. The dynamic behavior of various PCM codes is calculated and compared, e.g., asymptotically optimum PCM codes for DPSK, binary folded PCM with and without interleaving, asymptotically optimum PCM codes for coherent PSK, etc. We conclude that the best scheme for DPSK is simplified interleaving and an optimum PCM code for PSK. This paper also gives analytical formulas for double errorA-factors for standard PCM systems.     

Weighted Median Filters for Multichannel Signals

Weighted medians over multichannel signals are not uniquely defined. Due to its simplicity, Astola 's Vector Median (VM) has received considerable attention particularly in image processing applications. In this paper, we show that the VM and its direct extension the Weighted VM are limited as they do not fully utilize the cross-channel correlation. In fact, VM treats all sub-channel components independent of each other. By revisiting the principles of Maximum Likelihood estimation of location in a multivariate signal space, we propose two new and conceptually simple multichannel weighted median filters which can capture cross-channel information effectively. Their optimal filter derivations are also presented, followed by a series of simulations from color image denoising to array signal processing where the advantages of the new filtering structures are illustrated.     

Low Bit Rate Differential PCM for Monochrome Television Signals

This paper contains an examination of the performance of low bit rate differential PCM systems when used to encode monochrome National Television System Commission (NTSC) television pictures. 12 different encoders with various sampling rates and numbers of quantizing levels operating at bit rates from 9 to 24 Mb were considered. The differential PCM systems were implemented by using an 8-b PCM A/D converter followed by digital logic that performed the differential operation in a simple previous sample feedback loop. The subjective performance of these encoders was determined using an anchored seven-point quality rating scale and equipreference contours were plotted in a plane whose axes are sampling rate and quantizing bits. Entropy measurements were made on the output of the differential PCM encoders. These entropy measures were compared with the subjective performance of encoding systems operating at the same transmission rate. The encoding systems studied produced television pictures of medium quality in which quantizing noise and bandwidth limitations were apparent.     

Optimum Tapped Delay Line for Digital Signals

The problem of finding the optimum tapped delay line (TDL) gain settings for a digital communications receiver with a detection process involving a single threshold level (zero volts) is examined. Based on a minimax criterion which seeks to maximize the eye opening-to-rms noise ratio, an optimum set of tap gains is found with the aid of a geometric construction inn-dimensional space. An iterative procedure for calculating the tap gains is derived from then-dimensional space concepts. With some slight modifications, the same procedure is used to obtain the optimum TDL when the detection process involves two threshold levels(pm E volts). The optimization scheme described by Lucky is also given a geometric interpretation so that the two methods can be compared; in some cases, they yield identical results.     

语音信号的线性预测编码技术

本文以一个双声道声码器为例,来说明线性预测编码技术在处理语音信号中的应用.一个2400b/s,双声道线性预测编码的声码器,使用高速、可编程序的位片型微处理机,字长可扩充.并使用2K的RAM和1K的ROM的数据存储器.该系统设计成循环时间为208毫微秒的双总线结构.该声码器能同时处理两个声道,用微处理机和线性预测编码技术开发出该系统,并叙述使用该技术设计声码器的要领.     

中频PCM/FM信号全软化FM解调的一种实现方法

本文介绍了中频PCM/FM信号全软化FM解调系统的组成、解调原理及软件解调的一种实现方法,完成了对软件解调算法的理论阐述和数学推导,并根据推导出的数学模型在Matlab平台下进行软件设计,最后进行了仿真测试和实验验证,仿真结果和实验结果证明了该实现方法的可行性。     

语音信号的一种加权鲁棒特征参数W_RAS_MFCC

讨论了一种加权的带通滤波器分析法(WFBA)来加强语音信号相对自相关序列(RAS)美尔倒谱系数的性能,它通过简单而直接的手法使人耳敏感的对数带通滤波器(LFBEs)频谱峰值更加突出,而使相应不敏感的谷值处减小,实验表明,在噪声干扰或者有信道失真情况下,语音的相对自相关序列的加权Mel特征参数(W_RAS_MFCC)比加权Mel频率倒谱系数(W_MFCC)以及其他谱提升技术具有更强的鲁棒性.     

Time-Scale Modification of Speech Signals

This paper presents methods for independently modifying the time and pitch scale of acoustic signals, with an emphasis on speech signals. The algorithms developed here use parametric (sinusoidal) modeling techniques introduced by other authors, but new perspectives on the role of vocal tract decomposition and maintaining phase relationships between sinusoidal tracks are derived that achieve improved output quality with decreased computational load. Simulation results are provided to illustrate performance, and the algorithms developed here have been demonstrated capable of implementation on simple DSP hardware.     

具有延迟的欠定语音盲分离

探讨具有延迟的欠定语音盲分离问题,针对语音信号的部分W-分离正交性,提出一种新的基于混叠矩阵估计的语音盲分离方法.首先对观测信号进行短时傅里叶变换,选定一个信号作为参考信号,构造幅度衰减向量和时间延迟向量,提取单源主导区间的观测信号作为观测样本;然后对观测样本进行聚类,通过估计参考信号的混叠系数来估计混叠矩阵;最后结合...     

一种存在相干信号时的稳健的最优波束形成算法

提出了一种新的存在相干信号时的最优波束形成算法。该方法首先利用二阶统计量信息估计合成方向矩阵,然后给出了一种根据期望信号的波迭方向寻找与之对应的合成方向矢量的方法．再据此估计最优权矢量。该方法与现有的同类算法相比计算量更小、鲁棒性更强。计算机仿真证明了算法的性能。     

Optimum Finite Duration Nyquist Signals

This paper deals with the design of finite duration signals, which maximize spectral content inside a given band and which avoid intersymbol interference by being orthogonal to discretely delayed versions of itself. The signal is a solution of an integral equation With constraints. Techniques for numerically solving this equation are given. The signals are compared to the classical prolate spheroidal wave functions, for a few cases.     

Statistical Word Synchronization in PCM Transmission of TV Signals

Word synchronization in PCM of TV signals is more important than that of audio signals, since often a single channel of TV signals is transmitted independently. For this purpose, new word synchronization is proposed, where the statistical property of TV signals is utilized instead of adding a bit for synchronization. The property is that a more significant bit in PCM does not change word by word as frequently as a less significant bit, because of the correlation between adjacent picture elements, i.e., sampled values or words. Mathematical calculation shows that the average time between misframes is as long as 10⁴years and the average recovery time is less than several lines when a 4-bit reversible counter for synchronization is used in transmitting 500-kHz narrow-band TV-phone signals.     

Generic Weighted Filtering of Stochastic Signals

In this paper, a new theory of optimal weighted nonlinear filtering is presented. Two filter models are considered. The first model is based on a representation of the filter in the polynomial-like form with $q$ terms where each term consists of weighted matrices and the matrix determined from the error minimization problem. The second model extends the first one to the case of the filter concatenation. The filter models are given in terms of pseudo-inverse matrices, i.e., the requirement of invertibility for covariance matrices is omitted. Thus, our filters always exist. We develop methods which allow us to exploit advantages associated with the proposed nonlinear filter models. The methods consist of the orthogonalization procedure and the reduction of the original problem to $q$ individual minimization problems for smaller matrices. This leads to a considerable reduction in the required computational work. The error associated with the first filter model decreases when the number $q$ of terms of filter increases. Its compression ratio can be adjusted by varying a particular value of ranks in each of its $q$ terms. This means that the proposed filer structure provides the two degrees of freedom. The second filter model provides another degree of freedom, a number $k$ of filters in the concatenation. Variations of the degrees of freedom improve the performance of the proposed filters. In particular, the error associated with the filter concatenation decreases as the filter number $k$ in the concatenation increases.      

适合高阶QAM信号的加权多模盲均衡算法

该文提出了一种加权多模盲均衡算法。该算法结合了多模盲均衡算法和判决引导算法的各自优势,利用由判决符号的指数幂构成的加权项调整代价函数中的模值。在均衡器系数迭代过程中,加权项不仅随着判决符号自适应地改变,还可以根据MSE估计值作更精确地调整。理论分析和仿真结果表明,与多模盲均衡算法等其它算法相比,该文提出的算法在同等条件下可以获得更快的收敛速度和更低的稳态收敛残差,更适用于高阶QAM信号。     

Minimum Mean-Squared-Error Quantization in Speech PCM and DPCM Systems

The optimum (minimum mean-squared-error) and optimum uniform quantizing characteristics for Laplacian- and gamma-distributed signals are given in tabular form. The performance of these quantizers in real speech pulse-code modulation (PCM) and differential pulse-code modulation (DPCM) systems is investigated and compared with that obtained when the μ-law quantizing characteristic is used.     

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.     

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.     

语音信号的一种加权的鲁棒特征参数W_RAS_MFCC

本文讨论了一种加权的带通滤波器分析法(WFBA)来加强语音信号相对自相关序列(RAS)美尔倒谱系数的性能,它通过简单而直接的手法使人耳敏感的对数带通滤波器(LFBEs)频谱峰值更加突出,而使相应不敏感的谷值处更加减小,实验表明,在噪声干扰或者有信道失真情况下,语音的相对自相关序列的加权的美尔特征差数(W_RAS_MFCC)比W_MFCC以及其它的谱的提升技术具有更强的鲁棒性.     

线性预测语音数字的编码技术

以一个双声道声码为例，说明一性预测编码技术在处理语音信号中的应用，一个２４００ｂｉｔ／ｓ双声道线性预测编码的声码器使用高速、可编程序的片型微处理机、字长可扩充。并使用２Ｋ的ＲＡＭ和１Ｋ的ＲＯＭ的数存储器。该系统设计成物理 ２０８毫微秒的双一结构。该声码器能同时处理两个声道，用微处理线性预测编码技术开发出该系统，并叙述使用该技术设计怕码器的要邻。