基于TMS320VC5509的基音周期实时检测系统的设计

通过芯片TMS320VC5509和TLV320AIC23构建一个语音采集、数据处理的系统,实现了实时提取基音周期。在处理算法上,经过低通滤波之后对语音信号进行“三电平中心削波”处理,最后作自相关运算获取基音周期。实验结果表明,经过三电平中心削波能够很大程度地简化自相关函数的计算,从而有效和快速地提取基音周期。     

基于折线削波电平的自相关函数法检测基音

基音信息隐含于语音信号的局部极值点处,将语音信号分帧后,通过修正由局部极值点拟合的包络线得到折线削波电平,并将削波后的信号用自相关函数法估计基音。数值实验表明,语音信号经折线削波较中心削波后可保留更多的基音信息,并能有效地减少基音检测的错误。     

基于河南方言的低信噪比下基音检测算法研究

基音检测是河南方言语音信号处理中的一个重要环节,针对低信噪比环境下的河南方言语音基音检测准确率低的问题,提出了一种语音信号增强和基音检测相结合的算法.通过多窗谱估计的改进谱减法对语音信号进行降噪处理,对增强后的语音信号用中心削波法消除偏离基音轨迹的野点,再通过自相关法实现基音检测.仿真结果表明,对于低信噪比环境下河南方言语音信号的基音估值检测结果准确,估算出的基音频率和实际基音频率能很好的重合.     

舞蹈机器人中音乐基音频率的提取

舞蹈机器人能够跟随音乐的旋律和节奏作出不同的舞蹈动作,需要对音乐的特征参数进行实时有效的提取。基音频率是音乐的一个重要特征参数,基音频率提取的质量将直接影响音乐特征的识别效果。提出了一种基于自相关函数法的基音频率检测方法,先利用三电平中心削波法对信号进行削波,然后计算其自相关关系,为有效抑制谐波峰值干扰,再次计算信号的自相关性,进而提取音乐信号的基音频率。实验证明,该方法能够准确、稳定的提取音乐信号的基音频率,提取效果理想。     

基于Matlab的一种基音周期检测算法

在对自相关基音检测算法进行分析的基础上,对此算法进行了深入探讨,针对以往研究中存在的不足加以改进,考虑到检测准确度和检测速率两方面的因素,引入了三电平削波模块,设计了基于Matlab的估计方法,然后通过对一段具体的语音进行处理,得到了比较准确的浊音语音的基音周期.实验证明,该方法简单、有效.     

基于三阶累积量的语音信号的基音周期检测

本文提出了一种新的语音信号的基音周期检测方法,该方法根据语音信号的三阶累积量去确定语音信号的基音周期,能有效地排除白色或有色的高斯加性噪声所带来的干扰.与传统的基音周期估计的自相关函数法或平均幅度差函数法(AMDF)相比,该方法更精确、有效,具有更强的鲁棒性.     

一种适于计算声场景分析的混叠语音基音检测方法

本文提出了一种在混叠语音信号中检测各自语音分量基音信息的方法.该方法采用小波变换作为基音检测模型中的滤波处理,并用广义自相关运算突出基音信息,用增强自相关累和消除冗余信息,并提出了用基音概率函数来预测并跟踪不同基音的变化以提高基音检测的准确性.本文提出的方法可应用于计算声场景分析中.实验结果表明,该方法对于混叠语音的基音检测是非常有效的.     

基于matlab变声器的设计

为了实现由男声变换到女声,在语音信号参数分析过程采用短时自相关法提取语音信号的基音周期,同时用LPC倒谱分析法分析共振峰的范围,通过matlab编写程序修改语音参数并接近于女声的范围,构置GUI界面。在实验中,输入一段语音信号,输出时即实现了由男声到女声的变换效果。因此对于语音信号参数的修改能够实现男女声音之间的变换。     

基于小波变换的MBE模型基音估计算法改进

基音周期是语音信号最重要的参数之一。MBE模型的基音周期搜索算法存在着运算量大,抗噪性能一般等缺点。本文基于小波变换,对算法进行了改进。阐述了小波变换基音周期估计原理,基音估计算法实现和实验。分析和实验表明改进算法更具实用性。     

一种鲁棒的语音信号基音标注算法

阐述了一种连续语音信号的最高振幅位的基音标注算法,该算法是在声带振动信号中找到喉部振动关闭点,并进行参数调整,从而自动完成在连续语音信号中的基音位置标注。实验证明,该算法能有效地克服以往标注算法中基音周期估计时加倍或减半的误差,从而保证了基音标注的鲁棒性。     

基于小波变换的语音基频提取新算法

该文将小波变换应用于具有连续语音特征的三字词语音的基频提取,并针对实验中出现的问题对算法进行了改进,提出了一种新的基于小波变换的语音基频检测算法。该算法主要包括:离散小波变换计算、基于投票策略的基频点选择和基频起点确定、基频检查、异常点修正、头尾漏点处理以及基于投票策略的基频点精确定位。实验表明,该算法较好地克服了基于小波变换传统算法的不足,更适合于连续语音的基频提取,缺陷是需要较多的计算时间,不太适合于实时性要求较高的系统。     

Real-Time Implementation of Time Domain Harmonic Scaling of Speech for Rate Modification and Coding

Time domain harmonic scaling (TDHS) has been realized in real time on the Bell Laboratories digital signal processing (DSP) integrated circuit. It is an algorithm that can expand or compress the bandwidth and sampling rate of speech by taking advantage of the pitch structure in the speech signal. As such it is useful in a variety of speech applications including speech coding, speech enhancement, and rate modification. A single DSP can perform compression and a second DSP can perform expansion. Both operations require pitch information to be supplied with the input speech. Included in the system is a real-time pitch/periodicity detector which has also been implemented on a single DSP. Its design is based on a novel modification of the autocorrelation function type pitch detector. This paper presents details of both the TDHS and pitch detector implementation and discusses their performances. In particular in this paper we discuss a 2:1 compression and expansion system that has been used as part of a 9.6 kbit/s speech coder. TDHS was previously thought to require a much larger buffer than the RAM memory available in the DSP. We show that for all the compression/expansion ratios of interest the buffer size needed is twice the maximum pitch period.     

一种自相关基音检测算法

自相关基音检测算法是语音信号处理的关键技术,算法的效率直接影响了语音信号实时处理的质量。目前有许多较好的检测算法提高了基音检测算法的效率。而自相关基音检测算法是一种时域算法,它能直接对时域信号采样值进行分帧、求短时自相关函数,并根据一定的判决准则进行清浊音判断。提供了一种自相关基音检测算法,经过实验,它在保持较好的性能的基础上提高了检测效率。     

An integrated system for multichannel neuronal recording with spike/LFP separation, integrated A/D conversion and threshold detection

A mixed-signal front-end processor for multichannel neuronal recording is described. It receives 12 differential-input channels of implanted recording electrodes. A programmable cutoff High Pass Filter (HPF) blocks dc and low-frequency input drift at about 1 Hz. The signals are band-split at about 200 Hz to low-frequency Local Field Potential (LFP) and high-frequency spike data (SPK), which is band limited by a programmable-cutoff LPF, in a range of 8-13 kHz. Amplifier offsets are compensated by 5-bit calibration digital-to-analog converters (DACs). The SPK and LFP channels provide variable amplification rates of up to 5000 and 500, respectively. The analog signals are converted into 10-bit digital form, and streamed out over a serial digital bus at up to 8 Mbps. A threshold filter suppresses inactive portions of the signal and emits only spike segments of programmable length. A prototype has been fabricated on a 0.35-microm CMOS process and tested successfully, demonstrating a 3-microV noise level. Special interface system incorporating an embedded CPU core in a programmable logic device accompanied by real-time software has been developed to allow connectivity to a computer host.     

An autocorrelation pitch detector and voicing decision withconfidence measures developed for noise-corrupted speech

The authors describe an integrated speech feature extraction method consisting of: (1) a pitch detector; (2) a voicing decision to correctly partition speech into voiced and unvoiced intervals; (3) a confidence measure which reflects the probabilistic accuracy of the voicing decision; (4) a confidence measure which reflects the expected deviation of the pitch estimate from the true pitch and the probabilistic accuracy of this deviation; and (5) smoothing techniques for the pitch detector, the voicing decision, and the two confidence measures. The focus of their research is on voiced and unvoiced speech corrupted by high levels of white noise. The voicing decision and the confidence measures are developed by observing the behavior of three features derived from the autocorrelation function and experimentally fitting curves to the data. This integrated set of algorithms is statistically analyzed for speech at seven signal-to-noise ratios     

Estimating the pitch period of voiced speech

In speech processing an estimation of the speech pitch period is important. Real time pitch detection is only possible by the selection of an efficient algorithm suitable for implementation on a programmable processor or in special-purpose hardware. The use of the periodogram algorithm (p.a.) is proposed to detect the pitch period of voiced speech. This algorithm is attractive for the following reasons: (a) it has no multiply operation; (b) when implemented on a 16-bit computer (e.g. microprocessor) the computation can be done in integer arithmetic without exceeding the microprocessor's dynamic range; (c) it is a simple technique for estimating the pitch period with reasonable accuracy. Results of the analysis of speech signals and sinusoids using the periodogram algorithm are presented and comparisons are made with the average magnitude difference function (a.m.d.f.) which is an alternative method of estimating the pitch period of the voiced speech.     

广义时频表示的一种实时实现方法

本文研究了时频表示的实时实现算法及其算法优化问题，从广义局域自相关函数的物理意义出发，论证了广义时频表示的实时计算采用短时处理技术的可行性，并从两处着眼优化了算法，一得利用核函数共轭对称性，使时频表示的计算复杂度降低一半多；二是在计算解析信号过程中，给出了以任意帧移间隔的时域递推算法，进一步减少了实时算法的计算量。     

A Compact, Flexible LPC Vocoder Based on a Commercial Signal Processing Microcomputer

A very small, flexible, high-quality, full-duplex 2.4-kbit/s linear predictive vocoder has been implemented with commercially available integrated circuits. This fully digital realization is based on a distributed signal processing architecture employing three Nippon Electric Company (NEC) µPD7720 signal processing interface (SPI) single-chip microcomputers. One SPI implements the LPC analyzer, a second implements the Gold pitch and voicing decision algorithm, white the third µPD7720 implements the excitation generator and synthesizer. An Intel 8085-based 8-bit microcomputer is used for data transfer, control and multiplexing functions, and communications with the host terminal. The LPC chip set achieves high flexibility by accepting run time initialization options from the Intel 8085. These parameters include choice of linear predictive model (<= 15), analysis and synthesis frame size, and speech sampling frequency, as well as choice of speech input and output coding formats (linear or µ-255 law) and choice of analog or digjtal pre- and deemphasis. A total of 16 integrated circuits is used in the LPC vocoder with a power disipation of 5.5 W and occupying 18 in/sup 2/ of circuit area.     

遗传算法应用于超低副瓣线阵天线方向图综合

基于对标准遗传算法中收敛依赖于初始群体选择的困难所作的分析,提出交替使用两种遗传繁殖操作产生后代群体以摆脱收敛对初始群体选择的依赖.对于超低副瓣线阵天线的方向图综合问题,建立了改进的遗传算法优化模型.计算实例说明改进后的遗传算法其收敛不依赖于初始群体的选择,具有实际应用前景.     

Design and implementation of a robust pitch detector based on aparallel processing technique

The design and implementation of a parallel-processing-based pitch detector is presented. Pitch information is extracted by performing pitch detection on four different waveforms derived from the speech signal. Pitch information from the four pitch-detection processes is then combined to determine a final pitch estimate. The performance of this pitch detector is evaluated on a large database and compared to other well-known pitch detection algorithms. It has been implemented in real time on a TMS32020 fixed-point digital signal processor as part of a 2.4 kb/s vocoder. A performance comparison of the real-time fixed-point implementation and a computer simulation are also given. The results show that the pitch detector performance is maintained in the real-time implementation mainly because the majority of the algorithm computations are integer arithmetic and logic-type operations