一种基于小波变换的清浊音分类基音检测方法

提出了一种基于小波变换的鲁棒性基音周期检测方法。首先结合平均能量频带分布和短时过零率这两个特征参数对语音信号进行清浊音判决,然后对浊音段采用空域相关函数提取基音周期。实验表明,与传统的小波变换和自相关算法相比,该方法鲁棒性好,对基音检测具有更高的准确性。     

The time-domain periodogram algorithm

The Time-Domain Periodogram Algorithm (TDPA) is proposed for estimating the pitch period of voiced speech sounds. After the algorithm has been presented theoretically, experimental results of the analysis of speech signals and sinusoids using the TDPA are given and the algorithm's noise performance is also evaluated. These results compare favourably with the average magnitude difference function (AMDF). The TDPA is highly suitable for implementation on a 16-bit microprocessor, since it uses no multiply operations and the computation can be performed in integer arithmetic without exceeding the dynamic range of the microprocessor. The TDPA also provides as a byproduct a well behaved estimate of signal intensity.     

基于Teager能量算子和空域相关函数的基音检测算法

结合Teager能量算子和空域相关函数,提出了一种有效的基音周期检测方法。检测前在小波域上用Teager能量算子对语音信号进行清浊音判决,对浊音段采用空域相关函数提取基音周期。实验表明,与传统的小波变换算法和自相关法相比,该方法鲁棒性好,具有更高的准确性。     

基于小波变换的清浊音分类及基音周期检测算法

该文提出了一种基于小波变换的鲁棒性基音周期检测方法。检测前在小波域上用Teager能量算子对语音信号进行清浊音判决,对浊音段采用空域相关函数提取基音周期。实验表明,与传统的小波变换算法和自相关法相比,该方法鲁棒性好,具有更高的准确性。     

一种可用于CDMA移动通信的变速率语音编码算法

该文提出了一种码率为 0.75-5.4kb/s可变速率的高质量语音编码讲法。该算法对CELP的激励进行了改进,根据语音的特征把语音分成4类,不同类型的语音采用不同的激励码本。特别是对于浊音,提出了一种基于基音同步的嵌入分裂式激励码本,该码本利用浊音具有准周期性的特点,使该算法在很低的码率下就可很好地恢复浊音信号,克服了CELP在4kb/s速率以下因码本尺寸小而导致合成语音质量差的缺点。经非正式听音测试,它的主观质量超过了1～8kb/s的可变速率QCELP系统,并且平均速率大约只有2kb/s,比QCELP的5kb/s平均速率低了很多、非常适用于 CDMA移动通信系统。     

低速率WI编码器中4~6bit基音量化算法研究

基音在语音编码中通常采用7bit无失真均匀量化。由于浊音段语音的基音普遍具有缓慢渐变的特点,为了更有效地去除前后帧基音之间存在的相关性,该文基于Eriksson和Kang提出的4bit基音量化算法,针对汉语语音进行研究,实现了一套4~6bit基音量化算法。该算法计算简单,无需码书存储。将此基音量化方案应用于WI模型和WI编码器,主观A/B听力测试结果表明,该方案在高效量化基音的同时保证了合成语音质量几乎没有损失,完全满足低速率WI编码器对量化基音的要求。     

Unvoiced/voiced classification and voiced harmonic parameters estimation using the third-order statistics

Unvoiced/voiced classification of speech is a challenging problem especially under conditions of low signal-to-noise ratio or the non-white-stationary noise environment. To solve this problem, an algorithm for speech classification, and a technique for the estimation of pairwise magnitude frequency in voiced speech are proposed. By using third order spectrum of speech signal to remove noise, in this algorithm the least spectrum difference to get refined pitch and the max harmonic number is given. And this algorithm utilizes spectral envelope to estimate signal-to-noise ratio of speech harmonics. Speech classification, voicing probability, and harmonic parameters of the voiced frame can be obtained. Simulation results indicate that the proposed algorithm, under complicated background noise, especially Gaussian noise, can effectively classify speech in high accuracy for voicing probability and the voiced parameters.     

基于语谱图的语音端点检测算法

利用语音在语谱图中表现出的不同特征,提出了一种基于语谱图的语音端点检测算法。首先利用基音频率检测的原理在语谱图矩阵中搜索浊音段,然后计算出浊音段的信噪比,再根据信噪比和语谱图矩阵中浊音段的峰值进行完整的端点检测。因多数突发噪声并没有稳定的频率或者频率不在人的基音频率范围内,因此,该算法能够很好地抑制突发噪声的干扰,实验结果表明,在信噪比为10dB以上时该算法能够准确检测出语音的端点位置。     

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     

A Low-Rate Digital Formant Vocoder

A complete algorithm of a 1200-bits/s digital formant vocoder system is described. This vocoder algorithm draws heavily on the results of recent research in linear predictive coding. The transmitting parameters are frequencies and amplitudes of the first three formants, the pitch period, voiced/unvoiced decision, and the gain. Formant bandwidths are estimated at the synthesizer by using the amplitude information. The synthesizer structure is in the parallel form. The synthetic speech quality at 1200 bits/s is reasonably good; most of the speech is intelligible and speaker-recognizable.     

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。     

基于幅度压缩滤波的清浊音分类及基音估计

该文针对传统算法在实环境(不同噪声类型和信噪比)下容易发生清浊误判和基音估计错误问题,提出一种基于幅度压缩基音估计滤波(PEFAC)的清浊音分类及基音估计方法。首先,通过PEFAC削弱语音的低频噪声,提取出基音谐波;然后,采用基于对称平均幅度和函数的脉冲序列加权算法(SIM)确定谐波数目;最后,利用动态规划估计出基音,用基于3元素特征矢量的高斯混合模型对清浊音进行分类。仿真结果表明,在实环境下,所提方法能有效抑制清浊误判及基音估计错误现象的发生,性能优于传统方法。     

Influence of the position of the analysis frame in LPC pitch-synchronous analysis

The paper deals with the effect of the position of the time interval on the evaluation of voiced sound characteristics in speech analysis using the Linear Prediction Coding (LPC) technique. It is shown that when the analysis frame coincides with a single pitch period (pitch-synchronous analysis) an erroneous alignment (with respect to pitch pulses) of the analysis interval may introduce significant errors in the estimation of formant frequencies and bandwidths, and more generally, of sound spectrum. Synthetic speech is used to investigate the phenomenon and the experimental results are discussed. On the basis of these results, various techniques for reducing the influence of the position of the analysis interval on Linear Prediction parameters are discussed.     

Frequency-domain maximum likelihood pitch determination approach

The rate of oscillation of the vocal cords known as the pitch is an important sound feature that is useful in many speech applications. A novel approach for the automatic detection and estimation of the rate of oscillation of the vocal cords is described. The importance of this approach stems from the fact that pitch determination is conducted using three independent stages: a segmentation stage; a voiced-unvoiced classification stage; and a pitch estimation stage. Segmentation and the detection of voiced segments are implemented prior to pitch estimation in order to: exclude unvoiced sounds and silence from biasing the result of pitch estimation; employ a simple segmentation procedure with low computational complexity and time-delay; enhance the accuracy of voiced-unvoiced classification by including additional features in voicing detection; help pitch tracking by testing similarities over successive segments and to make use of a different analysis domain that enables a high resolution pitch estimation. A frequency-domain maximum likelihood procedure is used for the estimation of the pitch frequency of voiced segments by maximizing a log-likelihood function over the range of possible pitch frequencies in conversational speech. An efficient simplified realization of the generalized likelihood ratio segmentation method is also presented. Computer simulations on a number of utterances show that this approach gives an accurate, reliable and robust estimation of the pitch of voiced sounds.     

Research on the Speech Coding Algorithm at 1200bit/s

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Robust hybrid pitch detector

A hybrid pitch detector characterised by parallel analysis of the speech signal in temporal, spectral and cepstral domains is proposed. The voiced/unvoiced decision and pitch period evaluation is realised by a logical analysis of the results from three domains. The experimental analysis shows the robustness of the detector for noisy and telephone speech.<>     

基于小波变换的语音基音周期检测

语音信号是一种非平稳信号,基音周期是语音信号最重要的参数之一,传统的基音检测方法存在一些缺陷.小波变换鲁棒性强、能很好地反映信号的时频特性,非常适合处理非平稳信号.为准确提取基音频率,提出了一种基于小波变换的基音周期检测方法.检测前在小波域上用Teager能量算子分离出语音信号的浊音段,然后对浊音段采用空域相关函数降噪...     

汉语语音声调提取技术方法研究

随着现代科技和计算机以及平板电脑等的发展,语音交互将成为人机通信的主要方式,而汉语在语音合成中声调是不可或缺的一个重要组成部分。在声调提取过程中首先采用改进的短时自相关函数的方法进行基音检测,同时为了能较为精确地进行浊音的基音检测,利用变长分帧的方法提取基音周期序列,并通过Matlab仿真得到了汉语语音4种声调的调型曲线。仿真结果表明,该方法所得到的调型曲线与汉语普通话声调的典型曲线较为一致。     

一种改进的自相关基音检测算法

提出了一种改进的ACF基音检测算法。检测前在小波域上用Teager能量算子对语音信号进行清浊音判决,在基音检测过程的前端和末端加入了有效的预处理和后处理技术。实验结果表明,该算法比传统的自相关算法具有更高的准确性,在低信噪比下,基音周期提取和清浊判决具有令人满意的效果。     

Pitch synchronous differential predictive encoding system

A pitch synchronous differential predictive encoding system (p.s.d.p.e.) is described, which reduces the dynamic range of voiced speech to a value similar to that of unvoiced speech. As a consequence, the signal encoded has a smaller dynamic range than the speech signal and results in an improvement in the signal/noise ratio for a given transmitted number of bits per sample. This improvement is approximately 8 dB compared with an a.d.p.c.m. codec, when the p.s.d.p.e. system uses an adaptive p.c.m. encoder and the transmission rate is 3 bit/sample.