用于改变图像大小的M带插值滤波器设计

改变图像尺寸是多速率系统的重要应用之一。提出了一种通过采样率转换改变图像尺寸的有效方法，给出了线性相位FIR插值滤波器和线性相位FIR抗混叠滤波器的设计算法。结合实例说明了方法的有效性。     

全数字接收机中一种基于并行流水线与快速FIR算法的插值滤波器结构及其实现

该文在对已有的拉格朗日立方插值滤波器Farrow结构进行分析和研究的基础上,使用了流水线技术和并行处理技术来提高滤波器的速度。在此基础上提出了基于快速FIR算法的结构,降低了并行的Farrow结构的复杂度。对该算法结构进行了仿真,并在FPGA上实现。分析结果表明,改进后的结构有更快的运行速度和更低的功耗。     

一种基于并行处理技术的插值滤波算法及其FPGA实现

插值滤波器性能直接影响到全数字接收机的误码率,设计性能良好且易于硬件实现的插值滤波器是设计全数字接收机的关键.在对已有的拉格朗日立方插值滤波器Farrow结构进行分析和研究的基础上,使用了并行处理技术来提高滤波器的速度,并对该算法结构进行了仿真,在FPGA上实现.分析结果表明,改进后的结构有更快的运行速度和更低的功耗.     

浅谈数字音频采样率转换的插值算法

数字音频采样率转换的核心是插值算法,其决定了转换后音质损失的多少。本文介绍了音频A/D转换及采样率标准的基础知识,汇总了一些音频采样率转换的插值算法,并通过编写测试程序对音频数据进行采样率转换,将不同算法的计算耗时及插值结果进行比较后,总结各算法的优缺点及其适用场景。     

一种用于数字音频的插值滤波器设计

设计了一种应用于数字音频的插值滤波器。该滤波器采用多相插值原理,硬件电路包括并行数据输入接口、8倍插值器、16倍采样保持电路,实现对输入音频信号(PCM码)的128倍过采样。滤波器电路由VerilogHDL语言实现,利用SYNOPSYS提供的EDA工具进行仿真、综合,并通过FPGA验证,结果表明该电路能满足性能要求。     

软件无线电中改进CIC滤波器的研究

软件无线电接收机的灵活性之一是在于对调制方式没有限制。为此,不同采样率之间的转换对软件无线电终端来说就显得尤为重要。实现数字的采样率转换在传统方法中就是利用插值,而一般的传统方法不能满足反混叠效应的要求。给出了在采样率转换(SRC)中的一种改进级联积分梳齿(CIC)滤波器,从而更有效地避免了混叠效应。     

一种基于插值的定时同步算法研究

基于MSK调制通信系统,采用四阶差分定时误差检测算法对基于插值的定时同步环路进行了研究和分析,并提出了一种以插值输出信号均方误差最小为准则的插值滤波器设计,提高滤波器输出精度,改善定时同步环路同步性能。通过MATLAB仿真验证,与Lagrange插值滤波器相比,基于最小均方插值的定时同步算法,跟踪阶段定时抖动小,对噪声具有抑制作用,在信噪比较低的情况下也具有较好的同步性能。     

∑△DAC中插值滤波器的设计

本文设计了一种用于分辨率为20bit,采样率为48kHz,信噪比为102dB的∑△DAC(数模转换器)中的数字插值滤波器.利用多项插值器原理,采用128(插值,并利用SRAM和PLA设计了8倍插值,大大减少了所需硬件及芯片面积.仿真结果表明能够满足性能要求.     

数字滤波器在实际语音降噪中的应用

文章使用了2种类型的IIR数字滤波器,1种FIR数字滤波器用来处理人机交流中存在的两种噪声,高斯随机噪声以及正弦噪声。对一段纯净的语音同时加入这两种噪声,用3种基于不同方法设计的滤波器通过matlab仿真,输出不同滤波器滤波前后时域波形图以及频谱图。依据试验结果分析3种滤波器在两种噪声同时处理的效果。     

Adapted packet speech interpolation

Adapted packet speech interpolation (APSI) is presented as an evolution of digital speech interpolation (DSI) techniques. The inherent overload penalties of DSI are mitigated by the use of an overload strategy which distributes the penalties uniformly across all active speech sources. A novel use of linear delta modulation (LDM) allows the system to re-encode the input sources at various rates depending upon the total offered load to the system. The subjective performance of hardware is discussed. Two models of silence and talk-spurt behaviour (called activity) of speech are presented: an analytic model for single speakers obtained by the application of renewal theory, and a simulation model obtained from the analytic model.     

光滑不等距插值算法及其在遥测数据处理中的应用

根据部分遥测缓变参数列表校准数据的处理需求，提出了应用光滑不等距插值算法进行回归处理的方法。阐述了光滑不等距插值方法的原理，介绍了该方法的软件处理流程，并应用试验数据对算法进行了验证分析。该方法具有光滑性好及精度高的特点，根据得到的各区间上惟一的插值多项式可求出任一点的精确值，克服了全局多项式和样条插值方法的不足。试验结果表明：该方法适用于校准数据呈非线性各类缓变参数的译解处理。     

DAC中多级插值滤波器的研究与设计

设计了一种用于数字语音处理的插值滤波器。该滤波器利用多项插值原理,对采样率为44.1kHz,分辨率为16bit的输入音频信号（PCM码）进行128倍过采样。在Matlab下完成系统仿真后,采用Verilog对滤波器进行功能描述,并利用ModelSim工具进行仿真。仿真结果表明能够满足设计性能要求。     

Generalized polyphase structure-based Mth-band filter design and application to image interpolation

A generalized polyphase (GP) structure-based method for the design of linear-phase Mth-band FIR filters satisfying the time-domain interpolation condition is presented. Some new transform matrices that allow for the use of an arbitrary number of branches in the GP structure are developed. Closed-form frequency specifications for constituent filters of the GP realization are obtained, making an independent and fast design of the short-length constituent filters possible. The proposed GP-based method is then extended for the design of a class of interpolation filters with certain regularities. Some of the interpolation filters designed based on the proposed GP structure are also applied to image resizing. Simulation results show that our interpolation filter gives a superior resized image than the common cubic spline interpolation does.     

A frequency-weighted block LMS algorithm and its application to speech processing

In this letter, we propose a frequency-weighted block LMS (FWBLMS) algorithm that is based on minimization of the frequency-weighted block mean-squared error. The FWBLMS algorithm has an important advantage over the time-domain LMS algorithm in that it can be designed with different frequency weighting on error signal depending on the relative significance of various frequency bands. Application of the FWBLMS algorithm in adaptive linear prediction of speech is discussed.     

波形内插语音编码技术回顾与展望

介绍了原型波形内插和特征波形内插算法的基本原理及其实现方法，详细描述了原型波形内插和特征波形内插语音编码技术的研究进展情况，并提出了今后的研究方向。     

引入双边滤波器优化的彩色滤波阵列插值

基于彩色滤波阵列(CFA)的图像传感器在每个像素 位置获得三原色红(R)、绿(G)和蓝(B)中的一种分量 ,其缺失 分量需要根据周围像素插值得到。目前提出的许多种插值算法,绝大部分采用Bayer排列模 式。本文在色 差恒定假设基础上,提出一种基于双边滤波器的自适应Bayer模型插值算法,对G通道的估计采 用自适应滤波器进行插值,对R和B通道的插值采用双边滤波器。算法利用待插值像素 与不同距离像 素相关性不同的思想,根据图像边缘自适应设定滤波模板,能较准确估计G、B和R、G通 道之间的色差值。 实验结果表明,对比多尺度色差梯度算法和边缘强度滤波等算法,插值后的图像不仅 主观视觉, 且客观评价指标(彩色峰值信噪比,CPSNR)均优于这些算法。     

The smoothed pseudo Wigner distribution in speech processing

The Wigner distribution is shown to be a useful tool for the analysis of speech data. Cross-terms which characterize the Wigner distribution of multi-component signals have been suppressed using the smoothed pseudo Wigner estimator. This estimator possesses several advantages over the short time periodogram. Comparison of the SPWD to that of the spectrogram reveals that much better resolution is achieved by the SPWD.     

Audiovisual speech processing

We have reported activities in audiovisual speech processing, with emphasis on lip reading and lip synchronization. These research results have shown that, with lip reading, it is possible to enhance the reliability of audio speech recognition, which may result in a computer that can truly understand the user via hand-free natural spoken language even in a very noisy environments. Similarly, with lip synchronization, it is possible to render realistic talking heads with lip movements synchronized with the voice, which is very useful for human-computer interactions. We envision that in the near future, advancement in audiovisual speech processing will greatly increase the usability of computers. Once that happens, the cameras and the microphone may replace the keyboard and the mouse as better mechanisms for human-computer interaction     

JND's in the LPC poles of speech and their application toquantization of the LPC filter

For the purpose of guiding a pole quantization scheme, a psychophysical experiment was performed to measure just-noticeable differences (JND) in the frequency and radius of the poles. The frequency JNDs, measured up to a formant frequency of 4 kHz, are quantified as distributions with means that are increasing functions of formant frequency and bandwidth. An example of a pole quantization scheme, based on the JND data, is presented and found to be significantly superior to common scalar quantization methods of the LPC-PARCOR coefficients. The pole quantization scheme is found to be almost comparable, both in quality and bit consumption, to vector quantization