一种实用性强的MC-CDMA盲信道估计方法

基于子空间分解的MC-CDMA盲信道估计算法运算量大,实用性差.本文根据在MC-CDMA中信号子空间维数远小于噪声子空间维数的特点提出从信号子空间的角度进行MC-CDMA盲信道估计,并采用PASTd算法对信号子空间及其维数进行动态跟踪.本文还通过对子空间跟踪的结果进行正交化处理的方法,显著提高了盲信道估计的性能.本文的盲信道估计方法运算量低,实用性强,仿真结果表明其盲信道估计性能接近特征值分解法.     

基于信号子空间的语音增强方法

基于子空间的语音增强算法不同于基于信号处理和统计估计的经典语音增强算法,其核心思想就是将带噪语音信号映射到信号子空间和噪声子空间中,并在信号子空间中估计原始信号。本文提出的算法是以线性代数和矩阵分析为基础,利用对语音信号和噪声协方差矩阵同时对角变换的条件,对混有加性白噪声和粉红噪声的语音信号进行增强处理。经过实验分析及与传统的语音增强算法相比较,语音失真较小,增强效果较好,能够在极大限度地抑制背景噪声的同时减少频谱失真和残余噪声。     

一种基于信号子空间和听觉掩蔽效应的语音增强方法

采用子空间分解法对噪声语音进行信号增强，并用基于Johnston听觉掩蔽模型的感知滤波器对增强后的信号频谱进行平滑以抑制背景音乐噪声。几种噪声背景下对增强语音的客观测试表明，与传统的信号子空间分解法相比，基于信号子空间和听觉掩蔽效应的方法可有效地减少语音信号的失真度。     

Blind Adaptive Multiuser Detection for the MC‐CDMA Systems Using Orthogonalized Subspace Tracking

In this paper, we study the performance of subspace‐based multiuser detection techniques for multicarrier code‐division multiple access (MC‐CDMA) systems. We propose an improvement in the PASTd algorithm by cascading it with the classical Gram‐Schmidt procedure to orthonormalize the eigenvectors after their sequential extraction. The tracking of signal subspace using this algorithm, which we call OPASTd, has a faster convergence as the eigenvectors are orthonormalized at each discrete time sample. This improved PASTd algorithm is then used to implement the subspace blind adaptive multiuser detection for MC‐CDMA. We also show that, for multiuser detection, the complexity of the proposed scheme is lower than that of many other orthogonalization schemes found in the literature. Extensive simulation results are presented and discussed to demonstrate the performance of the proposed scheme.     

Integrated circuits for a real-time large-vocabulary continuousspeech recognition system

The architecture and implementation of a word processing subsystem for a real-time speech recognition system using hidden Markov models are described. The bottleneck of this system, which is the acquisition of data, is demonstrated, and an architecture that speeds up this bottleneck using on-chip dual-ported cache memories is presented. The architecture is described in a textual form, and the layout data were completely automatically generated. The chips have been fabricated through MOSIS using a 2-μm CMOS n-well technology. The functionality of the processors was successfully tested using the scan-path test methodology. The clock rate for the scan-path test was 5 MHz to guarantee proper operation of the circuits for this clock rate. All the processors were first time working silicon     

改进的基于信号子空间的多通道语音增强算法

通过同时对角化麦克风阵列接收信号中语音信号和噪声信号的全局协方差矩阵,本文改进了一种基于信号子空间分解的多通道语音增强算法.该算法不依赖任何信号模型且无需对噪声信号的统计特性进行任何先验假定,它弥补了原始算法只限于白噪声背景下语音增强的不足,实现了色噪声背景下语音信号的最优估计.仿真结果表明本文算法在主观和客观测试中都具有良好的语音增强效果.     

Systolic architectures for finite-state vector quantization

We present a new systolic architecture for implementing Finite State Vector Quantization in real-time for both speech and image data. This architecture is modular and has a very simple control flow. Only one processor is needed for speech compression. A linear array of processors is used for image compression; the number of processors needed is independent of the size of the image. We also present a simple architecture for converting line-scanned image data into the format required by this systolic architecture. Image data is processed at a rate of 1 pixel per clock cycle. An implementation at 31.5 MHz can quantize 1024×1024 pixel images at 30 frames/sec in real-time. We describe a VLSI implementation of these processors.     

基于子空间方法的最小二乘常模算法的研究

本文提出了两种基于子空间方法的常模算法,称为SUB_LSCMA和LSCMA_PASTd。SUB_LSCMA先采用奇异值分解(SVD)获得紧缩近似投影子空间(PASTd)算法的初值,用PASTd算法来计算信号子空间,并对该信号子空间作施密特正交化,将最小二乘常模算法(LSCMA)的权系数投影到正交的信号子空间上,目的是减轻噪声子空间干扰的影响,但复杂度比已有的基于直接对接收信号自相关矩阵做特征值分解(ED)的LSCM_SUB算法[6]复杂度低。LSCMA_PASTd在SUB_LSCMA的基础上作了进一步改进,采用改进的PASTd算法来计算信号子空间,该信号子空间具有正交性,并且对初值的选取不敏感,能运用于实际的多径衰落信道中。仿真结果表明这两种算法的收敛速度、跟踪性能和误码性能和LSCM_SUB算法基本相同,但是复杂度比LSCM_SUB算法低。     

Signal subspace approach for narrowband noise reduction in speech

A signal subspace method is proposed for speech enhancement in the presence of narrowband noise. A fundamental assumption in subspace methods for noise reduction is that the noise covariance matrix is positive definite. However, this is not always the case, especially when the noise has narrowband characteristics. Based on the eigenvalue decomposition of the rank deficient noise covariance matrix, it is shown how to formulate the enhancement algorithm by decomposing the vector space of noisy signal into a signal-plus-noise subspace and a noise-free subspace. The proposed subspace partition is different from the conventional subspace approaches in that the noise reduction algorithm is implemented using the whitening approach exclusively in the signal-plus-noise subspace. The enhancement is performed by estimating the clean speech from the signal-plus-noise subspace and adding the components in the noise-free subspace. An explicit form of the estimator is presented, and examples are illustrated to validate the effectiveness of the proposed method.     

一种基于信号子空间和听觉掩蔽效应的语音增强方法

提文推导了基于离散余弦变换(DCT)的子空间分解法对有色噪声背景下的语音进行增强的公式,用基于听觉掩蔽效应的感智滤波器对增强后的信号频谱进行平滑以抑制背景噪声。几种噪声背景下对增强语音的客观测试表明,本文提出的方法可以有效地减少语音信号的失真度。     

Iterative cyclic subspace tracking for blind adaptive multiuser detection in multirate CDMA systems

In this paper, the problem of subspace-based blind adaptive multiuser detection in multirate direct-sequence code-division multiple-access (DS-CDMA) systems adopting short (periodic) spreading codes is considered. The solution that we propose is based on the well-known formulation of the linear minimum mean-squared error and decorrelating detectors in terms of signal subspace parameters. Since in a multirate scenario the correlation properties of the observable and, hence, the signal subspace parameters are periodically time-varying, classical subspace tracking algorithms, which assume that the subspace to be tracked is time-invariant or slowly time-varying, are shown to be not useful in this situation. A new recursive cyclic subspace tracking algorithm is thus developed. This procedure, which is based on a generalization of the PASTd algorithm, is able to capture the periodical variations of the signal subspace, and thus enables subspace-based blind adaptive multiuser detection in multirate CDMA systems. The proposed algorithm has a smaller computational complexity than the recently developed cyclic recursive-least-squares procedure, and, as numerical results confirm, is capable of providing very satisfactory performance.     

基于人耳感知掩蔽效应的子空间语音增强算法研究

针对传统子空间语音增强算法中,因语音增强方法中去除噪声而出现的音乐噪声和失真问题,提出了一种人耳感知掩蔽效应的子空间语音增强算法,并结合频域到特征值域的变换,在Bark域内实现人耳的感知掩蔽效应的语音增强。实验结果表明,该算法在白噪声和有色噪声的背景下,与传统子空间语音增强算法相比,不仅提高了语音信号的信噪比,而且减少了语音失真和音乐噪声,提高了增强后语音的听觉质量。     

A Configurable Logic Based Architecture for Real-Time Continuous Speech Recognition Using Hidden Markov Models

An architecture is presented for real-time continuous speech recognition based on a modified hidden Markov model. The algorithm is adapted to the needs of continuous speech recognition by efficient encoding of the state space, and logarithmic encoding of the weights so that products can be computed as sums. The paper presents the algorithm and its application related modifications, the mapping of the algorithm to a special purpose architecture, and the detailed design of this architecture using configurable logic. Emphasis is given on how the attributes of the algorithm are exploited in a configurable logic based design. A concrete design example is presented with a coprocessor engine having one large FPGA, 64 Mbytes of synchronous DRAM (SDRAM), a small FPGA as a SDRAM controller, and 2 Mbytes SRAM. This engine operating at 66 MHz performs roughly nine times as fast as a high end personal computer running a fully optimized version of the same algorithm.     

一种基于ME算法的RS译码器VLSI高速实现方法

针对ME算法VLSI结构进行了分析,提出ME算法的流水线及最小化VLSI结构,以满足数据处理速率不断提高的需求。并利用该算法实现结构设计了一种低资源占用率、低成本的高速RS译码器。逻辑综合及仿真结果表明,基于Altera公司CycloneII系列FPGA的RS(255,239)译码器,工作时钟达210 MHz,可满足数据速率1.68 Gb·s^-1的编译码要求。     

色噪声下基于白化频谱重排鲁棒主成分分析的语音增强算法

基于鲁棒主成分分析(RPCA)的单通道语音增强算法是高斯白噪声环境下语音增强的一种重要处理手段,但其对低秩语音分量处理效果欠佳且无法较好地抑制色噪声。针对此问题,该文提出一种基于白化频谱重排RPCA的改进语音增强算法(WSRRPCA),通过优化噪声白化模型,将色噪声语音增强转换成白噪声语音信号处理,利用频谱重排改进RPCA语音增强处理算法,从而获得色噪声环境下语音信号处理性能的整体提升。仿真实验表明,该算法能够较好地实现色噪声环境下的语音增强,且相对于其他算法具有更佳的噪声抑制和语音质量提升能力。     

An improved subspace tracking algorithm for blind adaptive multiuser detection

As the Projection Approximation Subspace Tracking with deflation(PASTd) algorithm is sensitive to impulsive noise, an improved subspace tracking algorithm is proposed and applied to blind adaptive multi-user detection. Simulation results show that the improved PASTd algorithm not only remains the properties of the conventional PASTd algorithm, but also has good Bit Error Rate(BER) performance in impulsive noise environment, thus it can effectively improve the system performance.     

基于高速DSP的实时语音增强系统设计

随着移动通信技术的快速发展,语音增强的研究及其实际应用成为数字化通信的一个重要的研究方向。在数字信号处理技术的支撑下,许多优秀的语音增强算法的实时实现成为了可能。谱减法是一种运算量相对较小,增强效果明显,并且容易实时实现的语音增强算法,但是其缺点就是残留有音乐噪声。针对传统谱减法,本语音增强系统采用了一种改进算法,就是...     

Efficient hardware architectures for eigenvector and signal subspace estimation

We consider hardware solutions to the adaptive-signal subspace-estimation problem. In deriving a hardware-realizable subspace tracking algorithm, we have applied delayed updating to the PASTd algorithm to achieve high speed. Pipelined and systolic architectures and the estimation of the dominant eigenvector or the signal subspace are also studied. Methods for approximating a reciprocal computation are employed and simulation results are presented to validate our algorithm and hardware architectures.     

A Single Chip Digital Signal Processor and Its Application to Real-Time Speech Analysis

A single chip high-performance digital signal processor (HSP) has been developed for speech, telecommunication, and other applications. The HSP uses 3 µm CMOS technology and its architecture features floating point arithmetic and pipeline structure. By adoption of floating point arithmetic, data covering a wide dynamic range (up to 32 bits) can be manipulated. The input clock frequency is 16 MHz, and the instruction cycle time is 250 ns. Efficient signal processing instructions and a large internal memory (program ROM: 512 words; data RAM: 200 words; data ROM: 128 words) make it possible to construct a compact speech analysis circuit by the LPC (PARCOR) method with two HSP's. This paper describes HSP architecture, LSI design, and a speech analysis application.     

语音增强算法的研究与实现

语音增强算法也被称为噪声消除算法,在通信行业快速发展的现代社会,语音增强技术具有现实研究意义.首先介绍了算法的运行环境,即语音和噪声信号的特征.其次,对三种经典算法的原理、重要参数和运行框图进行了理论分析.最后以基于统计模型的语音增强算法作为例子进行仿真,验证了语音增强效果.