采用经验模态分解的语音与音频通用编码方法

为有效解决现有单一模型编码器无法在中低速率对语音和音频信号进行高质量通用编码的问题,本文借助语音与音频信号的谐波特性,建立了一种对语音和音频信号统一编码的方法。首先,本文利用经验模态分解(Empirical Mode Decomposition, EMD)提取输入信号的谐波成分;其次,利用感知匹配追踪算法,并结合正弦参数建模对谐波成分进行参数提取与量化;第三,对于量化谐波后的残差进行抖动格型矢量量化,以提升重建音频的主观听觉质量,并最终实现一套包含24kbps和32kbps码率的宽带语音与音频通用编码器;最后,对所提算法进行了客观PESQ/PEAQ和主观A/B测试,并与ITU-T G.722.1和G.722.2编码器进行了比较,实验结果表明,所提编码器对语音和音频信号的编码质量均优于参考编码器。      

一种面向高采样率音频信号的参数音频编码系统

通过对现有的各种参数音频编码方案进行比较和改进,提出了一种面向高采样率音频信号的参数音频编码系统,通过使用多分辨率正弦模型、多分辨率ERB噪声模型,辅以谐波结构分析、双变换短时成分检测与双变换短时模型,可在24Kb/(s·ch)的码率下实现对44.1kHz,48kHz采样音频信号的参数音频编码,主观音质良好。     

一种结合正弦参数提取和MDCT滤波器组的频谱优化方法

通过研究信号中正弦成分对改进的离散余弦变换(MDCT)的频谱产生的影响，针对使用MDCT滤波器组的音频编码系统，结合正弦参数提取，提出了一种对MDCT频谱进行优化的方法。这种方法利用MDCT和离散傅里里变换(DFT)之间的关系，在提取正弦参数的基础之上，通过简单线性变换求得MDCT谱，并在频域分离信号中的部分准静态正弦．从而达到优化MDCT频谱的目的。     

基于PCI总线的音频分析仪的设计与实现

本文介绍了基于PCI总线的音频分析仪的设计.该分析仪采用前端信号调理电路和高精度ADC+FPGA电路,通过PCI接口电路实现和主机的通信,能根据用户设定的波形类型、幅度、频率等信息产生相应的音频信号,能对采集到的音频数据进行特定的算法分析得到信号交直流电压、频率、谐波失真等参数.结果表明,本文设计的音频分析的音频信号分析频率精度高、总谐波失真(THD)低.     

参数音频编码算法的改进

参数音频编码是一种基于产生音频信号的参数模型的有效编码方法。最近已用于很低码率的音频编码中,如MPEG-4Version2音频标准HILN。这里提出HILN参数音频编码的改进算法。改进算法采用了自适应的基于倒谱的基频检测算法,利用倒谱和频谱特性,保证了大范围内基频提取的效率和准确性。在正弦波合成中采用相位匹配合成算法,保证了帧间参数的连续性。改进算法还对残差信号进行预处理,去除不必要的干扰分量,最大限度重建原音频信号,提高了合成音频信号的质量。     

基于GMM的甚低码率语音编码器

提出了一种新颖的基于高斯混合模型(GMM)的甚低码率语音编码系统.该编码器利用GMM对短时语音谱包络进行拟合的方法来对语音进行参数化表示.编码时,语音经预处理、分帧加窗后,再经FFT分析得到分帧语音的信号频谱,并获得平滑谱包络.然后采用GMM对谱包络进行拟合,用GMM参数(均值、方差、权重)对语音谱加以表示.由于GMM参数较少,从而可以使得码率甚低.解码时,根据编码逆运算生成谱包络,浊音信号利用正弦模型加以合成,清音信号经IFFT合成.实验仿真结果表明:该编码器在传输码率降低到2.35 kb/s时,仍可获得音质令人满意的解码语音.     

一种高精度改进型SHR基音检测算法

利用正弦语音模型中浊音存在的谐波与子谐波,在SHR(subharninctoharmonicratio)算法的基础上,提出了一种改进型高精度基音检测算法ISHR(improvingsubharninctoharmonicratio)。根据幅度调制和频率调制在语音分析中的特性、频域中幅度值和自相关频率比值,该方法采用基于正弦模型的均方误差对语音进行检测,提取出准确基音。仿真结果表明此种算法在基音提取中具有高精度及高可靠性。     

利用一种新的谱扩展技术提高窄带语音质量

提出了一种谱扩展算法,先对窄带语音进行线性预测编码(LPC)分析和多带激励(MBE)分析,得到窄带语音的线性谱频率(LSF)参数和清／浊音(U/V)判决信息;利用高斯混合模型(GMM)由窄带语音的LSF参数扩展得到高带语音的包络谱信息;对于激励信号,由窄带语音的U／V判决信息通过码本匹配得到高带的U／V判决,然后根据U／V判决信患由正弦模型合成出高带激励信号,并进而由LPC方法生成高带语音。最终的宽带语音由原始窄带语音和扩展得到的高带语音经子带合成得到.实验结果表明该算法扩展得到的宽带语音能有效提高语音信号的清晰度和可懂度。     

正弦参数自适应联合量化

为实现音频信号的低速率编码,提出一种正弦参数量化的新方案.该方案利用人耳掩蔽效应和实时参数统计特性,基于高速率原理建立并显式求解率失真优化问题,得到具有解析形式的参数量化器,实现了在正弦分量间和正弦参数间动态分配编码比特.本量化方案无迭代运算过程,适合实时低速率音频编码应用.与经典方法比较,平均比特速率每正弦减少约17%.当速率大于15bit/正弦,重建信号的感觉失真小于球形量化方案.     

基于总体最小二乘-旋转不变算法的地表核磁共振信号参数估计

在地表核磁共振(SNMR)找水系统中,根据SNMR信号的参数能够预估地下含水层的储水量、导电性以及孔隙结构等信息。然而在实际应用中探测现场采集的SNMR信号十分微弱,易受到环境噪声干扰,导致无法直接获取SNMR信号的参数。针对这一问题,该文提出基于总体最小二乘-旋转不变法(TLS-ESPRIT)的地表核磁共振信号参数估计方法。基于谐波噪声与SNMR信号的相似信号特征构成一个由多个正弦衰减信号叠加的混合信号模型,使用TLS-ESPRIT将混合信号参数提取问题转换为旋转不变矩阵的广义特征值求解,从而获得SNMR信号的拉莫尔频率和弛豫时间,并结合最小二乘法求得其初始振幅和相位。仿真信号和实测信号实验结果表明此方法能够估计出混有随机噪声和工频谐波噪声的SNMR信号的参数,相比传统的谐波建模方法,在参数提取精度上效果更好。     

Scalable perceptual audio representation with an adaptive three time-scale sinusoidal signal model

This work is concerned with the development and optimization of a signal model for scalable perceptual audio coding at low bit rates. A complementary two-part signal model consisting of Sines plus Noise (SN) is described. The paper presents essentially a fundamental enhancement to the sinusoidal modeling component. The enhancement involves an audio signal scheme based on carrying out overlap-add sinusoidal modeling at three successive time scales, large, medium, and small. The sinusoidal modeling is done in an analysis-by-synthesis overlapadd manner across the three scales by using a psychoacoustically weighted matching pursuits. The sinusoidal modeling residual at the first scale is passed to the smaller scales to allow for the modeling of various signal features at appropriate resolutions. This approach greatly helps to correct the pre-echo inherent in the sinusoidal model. This improves the perceptual audio quality upon our previous work of sinusoidal modeling while using the same number of sinusoids. The most obvious application for the SN model is in scalable, high fidelity audio coding and signal modification.     

On the use of a digital computer as a two-phase variable-amplitudevariable-frequency oscillator

A feasibility study on the use of a digital computer as a variable-amplitude variable-frequency oscillator which generates two-phase sinusoidal excitation control signals for a dual-excited synchronous generator is described. The oscillator uses two input signals. One input signal is used to control the frequency of the oscillator outputs, and the other is used to control their amplitudes. By a single variable, the software can be configured to vary the oscillator frequency range and to greatly reduce (almost eliminate) the harmonic distortion of the output signals. The harmonic distortion is constant and independent of the output frequency for each possible oscillator frequency range. The software-based oscillator design is flexible and can be used to generate different types of multiphase signal waveforms     

Frequency estimation of distorted non-stationary signals using complex H∞ filter

Mechanical vibration signals are always composed of harmonics of different order. A novel estimator is proposed for estimating the frequency of sinusoidal signals from measurements corrupted by White Gaussian noise with zero mean. Also low frequency sinusoidal signal is considered along with third and fifth order harmonics in presence of noise for estimating amplitudes and phases of different harmonics. The proposed estimator known as complex H_∞ filter is applied to a noisy sinusoidal signal model. State space modeling with two and three state variables is used for estimation of frequency in presence of white noise. Various comparisons in terms of simulation results for time varying frequency reveal that the proposed adaptive filter has significant improvement in noise rejection and estimation accuracy. Comparison in performance between two and three states modeling is presented in terms of mean square error (MSE) under different SNR conditions .The computer simulations clearly indicate that two states modeling based on Hilbert transform performs better than three states modeling under high noisy condition. Frequency estimation performance of the proposed filter is also being compared with extended complex Kalman filter (ECKF) under same noisy conditions through simulations.     

改进的移动音频带宽扩展算法

由于3G移动信道环境严酷,带宽资源有限,为了实现低码率、高质量的音频信号传 输和重建,在研究现有语音/音频带宽扩展技术的基础上,提出了一种基于EAAC+编码标准的 改进移动音频编码算法。该算法通过检测并去除短暂态信号带来的正弦分量,从而降低计算 的复杂度,实现传输信道带宽的扩展,并保证重建音频信号的音质。     

全光纤单芯语音传输系统及其保密性研究

提出了一种全新的全光纤语音传输方法。该方法利用光纤的光弹效应进行语音信号的录入,利用全光纤干涉系统进行信号的解调,成功实现了语音信号的长距离传输。该通信方式具有抗电磁干扰能力强、保密性好等优点。在对系统的工作原理及其保密性传输机理分析的基础上,通过实验验证了系统的T作原理,成功再现了300～3400Hz范围内的单频正弦信号及实际语音信号的录入、传输与还原过程,得到稳定的波形对比曲线及清晰的音质效果。该系统有望应用于距离为40km的点对点保密通信。     

基于提升小波的多重数字音频水印

该文提出了一种在提升小波域同时嵌入鲁棒水印和易损水印的音频水印算法。原始音频信号经过提升小波变换后,将低频小波系数进行均值量化嵌入鲁棒水印,具有较好的鲁棒性和不可感知性;对高频小波系数直接进行单个系数量化嵌入易损水印,当音频内容发生篡改时,这些水印信息会发生相应的改变,从而可以鉴定原始音频的完整性。水印的提取不需要原始音频信号。实验结果表明,鲁棒水印对MP3压缩、低通滤波、加噪、重量化、重采样等信号处理攻击具有很强的鲁棒性;而易损水印对上述攻击则具有很强的敏感性。     

改进的正弦模型在音频时域修正中的应用

介绍了一种更有效的相位内插算法,使正弦模型能够精确而更有效地分析/综合语音信号。将该正弦模型应用于音频的时域修正,通过使用正弦 瞬时冲击 噪声3种模型来灵活表示广泛的音频信号,分别针对不同的信号进行不同的时域修正,实验证明,通过此方法得到的效果明显优于传统时域修正算法。     

Damped and delayed sinusoidal model for transient signals

In this work, we present the Damped and Delayed Sinusoidal (DDS) model: a generalization of the sinusoidal model. This model takes into account an angular frequency, a damping factor, a phase, an amplitude, and a time-delay parameter for each component. Two algorithms are introduced for the DDS parameter estimation using a subband processing approach. Finally, we derive the Crame/spl acute/r-Rao Bound (CRB) expression for the DDS model and a simulation-based performance analysis in the context of a noisy fast time-varying synthetic signal and in the audio transient signal modeling context.     

Millimeter wave generation on nonlinear transmission lines

In recent years, nonlinear transmission lines (nltls) have successfully been used to generate ultrashort electrical pulses and transients from a sinusoidal input signal. In the frequency domain, this pulse compression results from the generation of a large number of harmonics with suitable phase relationships. In this paper, special examples of nltls are studied which are capable to produce millimeter wave signals at selected frequencies, here by second and third harmonic generation and nonlinear active wave propagation effects. Bi-modal transmission lines are particular discussed which provide the necessary dispersion for high conversion efficiencies.