A security watermark scheme used for digital speech forensics

Based on digital watermark, a speech forensics scheme is proposed. The feature coefficients cross-correlation degree of speech signal is defined, and the property is discussed, which demonstrates that the feature is very robust. Then a new watermark embedding method based on the feature is explored, aiming to enlarge the embedding capacity and solve the security issue of watermark schemes based on public features. In this paper, for each fame of speech signal, it is cut into two parts, and each part is divided into some segments. Then frame number is mapped to a sequence of integers, which are embedded into the segments. The integers can be extracted used for forensics and tamper location after watermarked signal being attacked. Theoretical analysis and experimental results show that the scheme proposed is inaudible and robust against desynchronization attacks, enhances the security of watermark system and has a good ability for speech forensics.     

基于压缩感知和MFCC的语音端点检测算法

在连续语音识别系统中,针对强噪声环境下传统双门限语音检测方法出现的误检问题,提出了一种结合压缩感知理论和MFCC倒谱系数的端点检测算法。该算法采用Hadamard随机观测矩阵和改进的OMP重构算法对语音信号进行压缩感知与重构,利用语音信号在离散余弦基上的近似稀疏性,提取重构信号的MFCC倒谱系数来检测语音信号的端点。仿真结果表明,提出的改进算法具有较强的鲁棒性,能满足在强噪声环境下对连续语音信号进行有效端点检测的要求。     

基于小波变换和压缩感知的语音信号压缩研究

对语音信号直接进行压缩感知处理,通常压缩的效率不高。针对此问题提出了一种基于压缩感知和小波变换的方法,首先用小波变换的方法对语音信号进行级数分解,然后采用压缩感知的方法对小波低频系数进行压缩,并丢弃高频系数,重构语音信号时高频系数用随机信号来取代。采用此种小波变换的方法,与直接采用压缩感知的方法相比,前者的语音信号MOS值稍有降低,但压缩率比直接压缩感知的方法降低了一倍,说明此方法可大大提高压缩的效率。     

小波变换的频响特性及其在语音去噪中的应用

讨论小波变换在实际语音信号去噪处理中应用。由于语音信号的复杂性 ,信号本身含有奇异性 ,因此不能单一使用阈值去噪法。文中定义了小波变换频响特性 ,并利用它重构低尺度参数上的小波变换模极大 ,达到去噪目的。实例证明它的有效性     

Content-Dependent Watermarking Scheme in Compressed Speech With Identifying Manner and Location of Attacks

As speech compression technologies have advanced, digital recording devices have become increasingly popular. However, data formats used in popular speech codecs are known a priori, such that compressed data can be modified easily via insertion, deletion, and replacement. This work proposes a content-dependent watermarking scheme suitable for codebook-excited linear prediction (CELP)-based speech codec that ensures the integrity of compressed speech data. Speech data are initially partitioned into many groups, each of which includes multiple speech frames. The watermark embedded in each frame is then generated according to the line spectrum frequency (LSF) feature in the current frame, the pitch extracted from the succeeding frame, the watermark embedded in the preceding frame, and the group index which is determined by the location of the current frame. Finally, some of the least significant bits (LSBs) of the indices indicating the excitation pulse positions or excitation vectors are substituted for the watermark. Conventional watermarking schemes can only detect whether compressed speech data are intact. They cannot determine where compressed speech data are altered by insertion, deletion, or replacement, whereas the proposed scheme can. Experiments established that the proposed scheme used in the G.723.1 6.3 kb/s speech codecs embeds 12 bits in each compressed speech frame with 189 bits, and only decreases the perceptual evaluation of speech quality (PESQ) by 0.11. Additionally, its accuracy in detecting the locations of attacked frames is very high, with only two normal frames mistaken as attacked frames. Therefore, the proposed watermarking scheme effectively ensures the integrity of compressed speech data.     

压缩感知分组分离语音增强

压缩感知（Compressive Sensing,CS）是一种基于信号稀疏性的采样方法,可以有效提取信号中所包含的信息。提出了一种分组分离压缩感知语音增强新算法。算法利用语音在离散快速傅里叶变换（Fast Fourier Transform,FFT）域下的稀疏性,设计复域观测矩阵与软阈值对带噪语音进行压缩测量与去噪,通过可分组分离逼近稀疏重建（Sparse Reconstruction by Separable Approximation,SpaRSA）算法恢复语音信号,实现语音增强。实验表明：该算法对含噪信号压缩重构,信噪比幅度较大提高,能更有效地抑制背景噪声。     

基于压缩感知的电力系统故障选线研究

为了解决电力系统故障选线中信号的采样、传输和存储问题,提出了一种全新的基于压缩感知理论的信号压缩的方法.该方法的采样频率不用考虑奈奎斯特采样频率.采样的信号是有选择性的部分信号.并通过设计重构算法来准确恢复该全部信号.考虑到一般条件下信号稀疏度不确定性,采用一种分割增广拉格朗日收缩算法(SALSA)来重构这些稀疏度不确定的信号.通过采用快速傅里叶变换基与高斯随机矩阵并且和SALSA相结合能够很好地实现信号压缩重构.对重构信号采用小波分解,获取重构信号的主要特征,分析零序电流模极大值的极性,找出其中一条与另外两条零序电流模极大值极性不同的线路,从而确定此线路为故障线路.     

基于言语情境分析的数字语音篡改检测

针对使用拼接手段的数字语音篡改,提出一种基于言语情境分析的篡改检测方法。该方法从背景噪声分析和说话人状态特征分析两方面入手,把语音信号分为语音部分和静音部分,对包含噪声的各个静音片段各帧提取时域和频域特征,对各语音片段提取韵律特征和音质特征,并分别基于贝叶斯信息准则检测特征的跳变点,通过综合判断得到篡改检测结果。实验结果表明,该方法能够比较准确地检测和定位语音拼接点。     

二次离散小波变换的语音增强方法

提出了一种基于二次离散小波变换（DWT）的语音增强算法。该算法首先对带噪语音信号进行离散小波变换,提取离散细节信号,并对其进行第二次离散小波变换。再按照不同的规则选取阈值,对信号进行去噪处理。最后再对出来后的语音信号进行合并。对比实验结果表明,该方法具有良好的消除噪声的效果,提高了语音的清晰度和可懂度。     

A speech content authentication algorithm based on a novel watermarking method

Some audio watermark schemes robust against desynchronization attacks are based on synchronization code embedded by quantifying signal energy, which have some shortcomings. Such as, (1) they do not verify the authenticity of watermarked signal detected. (2) They are vulnerable to substitution attack. To address the shortcomings and considering the background, a speech content authentication algorithm is proposed in this paper. Firstly, the original speech signal is framed, and each frame is cut into some segments. Secondly, samples of the segments are scrambled, and self-correlation of the scrambled signal is calculated. Lastly, watermark bit generated by frame number is embedded by quantifying the self-correlation. If watermarked signal is attacked, the attacked frames can be detected according to the frame number extracted. Theoretical analysis and experiments demonstrate that the scheme is robust against desynchronization attacks, improves the security, and has a good performance in ability of tampering location.     

Speech transmission with COFDM based on different discrete transforms

Recently, the multimedia and cellular technologies have spread dramatically. Therefore, the demand for digital information has increased. Speech compression is one of the most effective forms of communication. This paper presents three approaches for the transmission of compressed speech signals over convolutional Coded Orthogonal Frequency Division Multiplexing (COFDM) system with a chaotic interleavering technique. The speech signal has is compressed using the Set Partitioning In Hierarchical trees (SPIHT) algorithm, which is an improved version of EZW and which is characterized by a simple and effective method for further compression. For mitigation of the fading due to multipath wireless channels, this paper proposes a COFDM system based on fractional Fourier transform (FrFT), a COFDM system based on discrete Cosine transform (DCT), and a COFDM system based on discrete wavelet transform (DWT). The FrFT has the ability of solving the frequency offset problem, which causes the received frequency-domain sub-carriers to be shifted, and therefore, the orthogonality between subcarriers deteriorates even with equalization. The DCT has an advantage of increased computational speed as only real calculations are required. The DWT is spectrally efficient since it does not utilize cyclic prefix (CP). These systems have been designed under the assumption that corruptive background noises are absent. Therefore, denoising techniques, namely wavelet denoising and Wiener filtering methods are suggested at the receiver to achieve enhancement in the speech quality. The simulation experiments shows that the proposed COFDM–DWT with Wiener filtering at the receiver has a better trade-off between BER, spectral efficiency and signal distortion. Hence, the BER performance is improved with small bandwidth occupancy. Moreover, due to the denoising stage, the speech quality is improved to achieve good intelligibility.     

Asymmetric self-recovery oriented stereo image watermarking method for three dimensional video system

To authenticate integrity of the stereo image for three dimensional video systems, a new asymmetric self-recovery oriented stereo image watermarking method is proposed by considering inter-correlations between the left and right views of the stereo image. An asymmetric self-recovery mechanism is presented to conduct allocation of watermarking capacity for asymmetrically embedding recovery references of each view. The presented mechanism is also used to recover tampered left and right views asymmetrically to obtain high quality of the stereo image with the help of disparity when the tampered regions of the stereo image are authenticated. For security, a chaotic function is employed to generate authentication bits to detect tamper. The high-frequency energy of discrete wavelet transform is utilized to divide blocks of each view into smooth and complex types, so that the recovery reference with alterable bits is computed. Moreover, to obtain a trade-off between transparency and watermarking capacity, human visual characteristics are taken into account and then a just-noticeable difference model is exploited to classify the blocks into sensitive and insensitive types, which defines two or three least significant bits of pixels are allocated for embedding watermark. Experimental results demonstrate that the proposed method can reconstruct tamper efficiently and outperform other stereo image watermarking methods, especially for extensive tamper.     

基于过完备字典的体域网压缩感知心电重构

针对体域网远程监护中心对重构的心电信号（Electrocardiogram,ECG）精度要求高和体域网（Body sensor network,BSN）低功耗问题,提出基于过完备字典的体域网压缩感知心电重构方法. 该方法利用压缩感知理论,在传感节点端利用随机二进制矩阵对心电信号进行观测,观测值被传送至远程监护中心后,再利用基于K-SVD算法训练得到的过完备字典和块稀疏贝叶斯学习重构算法对心电信号进行重构. 仿真结果表明,当心电信号压缩率在70%～95%时,基于K-SVD过完备字典比基于离散余弦变换基的压缩感知心电重构信噪比高出5～22dB. 该方法具有信号重构精度高、功耗低和易于硬件实现的优点.     

基于自相关和Hough变换的丢失同步信息的视频图像还原

随机视频信号是指同步信息不明确的视频信号,其同步信号的频率和格式未知或随时间变化。同步信息的缺失对随机视频图像的还原造成了一定的困难。针对计算机视频信号简单分析了其组成及视频帧结构。在缺失同步信息的情况下,基于视频图像行与行之间的强相关性,提出了基于自相关的视频图像还原方法,并在视频图像基本还原的情况下利用Hough变换的方法对图像进行了校正优化。通过实测数据实验,结果表明,提出的方法能够有效、可靠地还原随机视频图像,实验结果对随机视频信号处理和应用具有重要意义。     

一种基于Hilbert-Huang变换的基音周期检测新方法

利用Hilbert-Huang变换对语言信号处理中基于事件的基音周期检测问题提出了一种新的检测方法．该方法利用Huang等人1998年提出的具有高时频分辨能力的Hilbert-Huang变换分析语音信号，并提取其瞬时能量，通过精确定位声门脉冲发生的时刻，从而精确地跟踪基音周期的变化，达到精确检测基音周期的目的．与传统方法相比，其优点主要表现在：（1）不需要对语音信号作短时平稳性假设；（2）检测精度高，适应范围广；（3）具有跟踪基音周期变化的能力；（4）能精确区分清浊音}（5）与传统方法相比，帧长大大增加，因而，在提取连续语音信号的基音轮廓时，用于分帧和拼合的开销大大减少，帧间拼合痕迹小．仿真数据和实际语音信号检测实验均获得了相当精确的检测结果．最后，需要指出的是，Hilbert-Huang变换作为一种新的信号分析方法，被成功地用于提取语音信号的基音周期，这本身是一个有意义的探索，它为拓展Hilbert-Huang变换理论的应用给出了一个新的尝试．     

基于子带分解的DFRFT自适应滤波语音增强算法

提出一种改进的语音增强方法,利用子带分解对带噪语音信号进行处理,再在离散分数傅里叶变换(DFRFT)域采用最小均方(LMs)自适应算法进行滤波,对滤波后的子带信号进行DFRFT逆变换,最后利用综合滤波器组合成增强后的语音信号。仿真结果表明,本算法明显提高了收敛速度,减少了计算时间。在主客观评价中均具有较好的语音增强效果。     

小波消噪和ICA在语音信号分离中的应用

为了消除语音信号分离中仍存在的部分混叠声音,提出一种基于小波消噪和独立分量分析（ICA）结合的信号分离方法。该方法将小波变换和独立分量分析结合,利用小波变换的去噪作用,滤除原始语音信号中的噪声后作为ICA的输入信号,采用FastICA算法在小波域进行独立分量分析,对输入信号实施分离。实验结果表明,该方法大大调高了传统独立分量分析对语音信号的分离效果。     

基于FFT的非均匀重采样值的重构实现技术

文章叙述数字语音信号的基于FFT的非均匀重采样值重构的技术实现问题,并对实际的WAV格式的数字语音文件的重采样值用VC++6.0编写的程序实现了基于FFT的重构;以实验数据结果分析讨论了重构信号的质量、重采样后的数据量与分段长度的关系等问题;实现方法、实现程序及结论不仅对WAV格式的数字语音文件有效,而且也适用于其它格式的数字语音文件和非语音信号的非均匀采样值重构的实现。     

基于自适应加权谱内插的宽带语音编码算法

提出了一种基于自适应加权谱内插(STRAIGHT)的宽带语音编码算法。输入的语音信号首先经过STRAIGHT分析得到精确的基频参数和谱参数，然后通过时域抽取和频域建模实现有效的编码压缩。在时域抽取时采用的区别于传统编码算法固定帧长的自适应可变帧长方法，使得编码存储量可以根据实际语音变化情况得到更加合理的分配。主观测听结果表明，该算法针对16kHz采样的语音信号，在6kbps码率上可以取得与AMR-WB(G．722．2)在8．85kbps时的相当的音质效果。此外，该算法还具有对恢复语音的时长、基频以及谱参数较强的调整能力。     

叠加特征信息辅助的语音传输与重构

为改善压缩语音传输系统的重构精度且不增加系统的频谱开销,提出一种叠加特征信息辅助的语音压缩传输与重构方法。提出方法首先提取稀疏语音信号的特征信息;抽取的特征信息以叠加序列方式叠加在压缩语音信号上进行传输;接收机重构时,借助特征信息辅助重构算法进行语音重构。分析与仿真结果表明,相比于传统的压缩感知语音重构方法,在较高信噪比或较低压缩率情况下,提出方法可改善语音重构精度,且不增加传输系统的频谱开销。