结合EMD与DWT-ACF的语音基音周期检测改进算法

针对传统小波-自相关算法在噪声环境下检测语音的基音周期会出现偏差和漏报的情况,提出一种经验模式分解下的小波-自相关的基音周期检测改进算法。该算法首先利用经验模式分解去除含噪语音趋势项并减噪,再利用改进的小波-自相关法突出每个基音周期的峰值点,提高了基音周期检测的精度。实验结果表明,该改进方法可有效改善加噪语音在基音提取上出现的偏差误报情况以及避免部分倍频和半频错误,提高基音周期检测速率及准确率。     

基于趋势分析的AMDF基音检测改进算法

针对语音信号处理中传统AMDF基音检测算法存在下降趋势易出现检测错误问题,提出基于去均值下降趋势的AMDF算法改进框架,并以线性多项式表示AMDF均值下降趋势,给出基于最小二乘法的AMDF改进算法。利用最小二乘法拟合均值下降趋势;将其从AMDF中减去获得改进的AMDF(LSAMDF)。仿真实验表明LSAMDF性能明显优于多种AMDF改进算法,验证了改进框架的正确性及以线性趋势表示均值下降趋势的合理性。     

MELP声码器的算法研究及实现

为了满足数字通信及其它商业应用的需求,语音压缩编码技术得到了迅速发展。特别是低码率语音编码的研究具有十分重要的现实意义。在现有的语音编码研究中,混合激励线性预测编码(MELP)是一种比较好的方法。对MELP编解码算法的原理进行简要分析,讨论如何在MATLAB上实现该算法,并研究其关键技术,最后对测试结果进行分析和比较。     

用同态解卷估计褐稻虱鸣声的基音周期

本文介绍同态解卷的基本原理以及它在褐稻虱鸣声分析中的应用.扼要介绍了同态滤波.提出了褐稻虱发声的机理和褐稻虱鸣声产生的声学模型.根据这一模型.用实倒谱区分鸣声属于“浊音”还是”清音“.如果属于“浊音”.就估计它的基音周期.最后,测出了在一声鸣叫中基音周期的变化.     

超低速率MELP语音编码算法研究

在语音编码算法中,混和激励线性预测(MELP)算法因为能更好的模拟自然语言特征,在低速率上能合成较高质量的语音,而成为现代低速率语音编码中最有潜力的算法之一。但在无线通信、卫星通信以及军用和保密通信中,信道带宽成为一个突出的问题,因此对更低速率语音压缩编码技术乃至超低速率的语音压缩编码技术的研究是非常有必要的。针对语音通信中关于极低速率的要求,深入分析了现今的几种基于MELP的低速率语音编码算法,对其原理以及关键技术进行了归纳总结,并对语音质量进行了比较。     

An effective codebook search algorithm for vector quantization

Abstract

A new scheme that aims to cut down on the computational cost of the vector quantization (VQ) encoding procedure is proposed in this paper. In this scheme, the correlation between the codewords in the codebook is exploited and three test conditions are designed to filter out the impossible codewords in the codebook. The design of test conditions is based on the concept of integral projection.

From the experimental results, it is shown that the new scheme outperforms all the other schemes proposed so far in speeding up the VQ encoding procedure. When the codebook of 1024 codewords is used in the proposed scheme, the execution time it consumes is less than 2 per cent of that needed by the full search algorithm. The average time reduction rate is approximately 97.7 per cent compared to the execution time for the full search algorithm. In other words, the proposed scheme indeed provides an effective approach to speed up the VQ encoding procedure.     

Classified vector quantization of LPC parameters

Abstract

To achieve high coding efficiency, modern speech coders adopt hybrid coding approaches, which utilize different coding mechanisms for various classified speech segments. With known voiced/unvoiced detection, in this paper, a classified LPC quantization (CLPQ) scheme is presented to effectively encode line spectral frequencies (LSF). The proposed CLPQ scheme improves the performance of the classified LSF vector quantizer, which adopts two LSF codebooks derived separately from voiced and unvoiced speech frames. With an objective spectral distortion measure, the CLPQ scheme successfully reduces the bit rate by about 1 bit/frame. Many classified LSF quantizers with different codebook structures and bit rates were evaluated. It would be helpful to design a classified LSF quantizer, which arrives at a compromise between distortion, bit rate and computational complexity.     

Wavelet‐based adaptive vector quantization for still‐image coding

Wavelet transform coding (WTC) with vector quantization (VQ) has been shown to be efficient in the application of image compression. An adaptive vector quantization coding scheme with the Gold‐Washing dynamic codebook‐refining mechanism in the wavelet domain, called symmetric wavelet transform‐based adaptive vector quantization (SWT‐GW‐AVQ), is proposed for still‐image coding in this article. The experimental results show that the GW codebook‐refining mechanism working in the wavelet domain rather than the spatial domain is very efficient, and the SWT‐GW‐AVQ coding scheme may improve the peak signal‐to‐noise ratio (PSNR) of the reconstructed images with a lower encoding time. © 2002 Wiley Periodicals, Inc. Int J Imaging Syst Technol 12, 166–174, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ima.10024     

基于量化步长线性预测和BFOS算法的MPEG-4AAC量化

MPEG-4 AAC的编码性能很大程度上依赖于量化模块的编码效率和收敛速度,但其常用的基于双循环搜索结构的率失真控制器引起编码器性能较差,尤其在低码率时更为突出。提出一种新的量化优化算法。新方案采取单循环结构,用前面数帧的量化信息对当前帧的初始量化步长做线性预测,再用接近最优比特分配的BFOS算法控制量化步长的调节。仿真证明新方案的编码性能明显优于MPEG-4 AAC VM,对比BOFS算法,运算量得到极大降低。     

Particle filtering based pitch sequence correction for monaural speech segregation

This article proposes a novel speech and sound segregation framework incorporating a technique for correcting a series of pitch periods based on particle filtering. The conventional pitch track correction method finds the peak locations of the autocorrelation functions to estimate the pitch period, and only the longest reliable pitch streak is used to correct unreliable pitch tracks. Especially in noisy environments, it is hard to find long and reliable pitch streaks, resulting in the degradation of the speech segregation performance. The proposed algorithm based on particle filtering considers all the reliable pitch streaks rather than the longest one and smoothly connects the scattered pitch streaks. To apply the particle filtering algorithm to pitch track correction, the importance weight computation to account for the degree of matchness of the found pitch to the individual spectro‐temporal components is also proposed. The performance of the proposed method is evaluated by the results of speech segregation experiments for the mixtures of speech and various noise sources in various mixing signal‐to‐noise ratios (SNRs). The evaluation measures were SNR, energy loss ratio, and noise residue ratio of the segregated speech, and all these measures showed that the proposed segregation method achieved superior performance compared to the conventional approach. © 2013 Wiley Periodicals, Inc. Int J Imaging Syst Technol, 23, 64–70, 2013.     

On the fast search algorithms for vector quantization encoding

One of the major difficulties arising in vector quantization (VQ) is high encoding time complexity. Based on the well‐known partial distance search (PDS) method and a special order of codewords in VQ codebook, two simple and efficient methods are introduced in fast full search vector quantization to reduce encoding time complexity. The exploitation of the “move‐to‐front” method, which may get a smaller distortion as early as possible, combined with the PDS algorithm, is shown to improve the encoding efficiency of the PDS method. Because of the feature of energy compaction in DCT domain, search in DCT domain codebook may be further speeded up. The experimental results show that our fast algorithms may significantly reduce search time of VQ encoding. © 2003 Wiley Periodicals, Inc. Int J Imaging Syst Technol 12, 204–210, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ima.10030     

基于相似度的高精度基音检测算法

提出了一种具有较高精度且抗噪性能强的基音检测算法。该算法将线性预测残差看作语音源信号的近似,对其进行频谱分析,依据残差幅度谱算得基音周期的粗估值。然后回到时域信号,根据基音周期粗估值设计一长度可调的窗,通过窗函数在语音段连续取两段语音信号作相似度运算,可根据最大相似度值计算出准确的基音周期。该方法准确性高,在噪声环境下也具有较好的效果。     

一种改进的基于对数谱估计的语音增强算法

提出了一种基于对数谱估计的改进型语音增强算法。相对于传统语音增强算法,在语音信号存在不确定的条件下,利用软判决增益因子修正技术调正带噪语音信号的对数谱幅度,抑制背景噪声。引入的改进型先验信噪比估计和语音信号先验不存在概率估计方法,能够有效地估计得出语音信号的存在概率,进而求得语音信号存在时的谱增益因子函数,联合语音信号不存在时设定的增益因子函数加权求得谱增益函数。计算机仿真表明,即使在低信噪比条件下,输入背景噪声为高斯白噪声和粉红噪声等加性白噪声时,所提算法对噪声的抑制效果非常明显,且有效地克服了传统算法中引入的“音乐噪声”和语音信号畸变。     

一种新的自适应量化数字音频水印算法

文章中提出了一种新的自适应量化数字音频水印算法，该算法首先将视觉可辨的二值水印图像降维成一维水印序列，并对水印序列进行随机置乱与BCH纠错编码，再将原始数字音频信号划分成音频数据段，最后选择音频段进行快速傅立叶变换(FFT)，并依据人类听觉系统(HAS)模型自适应确定量化步长量化FFT系数嵌入水印信息。该算法提取水印信息时不需要原始数字音频信号。仿真结果表明：该自适应量化数字音频水印算法不仅具有较好的透明性，而且对诸如叠加噪声、有损压缩、低通滤波、重新采样等攻击均具有较好的鲁棒性。     

Iterative optimization for joint design of source and channel codes using genetic algorithms

Abstract

This paper presents a novel algorithm for the joint design of source and channel codes. In the algorithm, channel‐optimized vector quantization (COVQ) and rate‐punctured convolutional coding (RCPC) are used for design of the source code and the channel code, respectively. We employ the genetic algorithm (GA) to prevent the design of COVQ from falling into a poor local optimum. We also adopt the GA to reduce the computational time needed for realizing the unequal error protection scheme best matched to the COVQ. Both the GA‐based source coding and channel coding scheme are then iteratively combined to achieve a near global optimal solution for the joint design. Numerical results show that the algorithm can be an effective alternative for applications where high rate‐distortion performance and low computational complexity are desired.