一种多带清浊音判决方法

文章提出一种根据ＬＰＣ 语音残差信号在频域的子带波形,计算其自相关函数,分析每个子带的周期性和非周期性。进行清、浊音判决的方法,可有效地降低 ＭＢＥＬＰＣ 算法中ＵＶ 判决的计算复杂程度,减少计算所需的存储空间。该判决方法与全带语音ＵＶ 判决的原理一致。经实际听音试验,这种判决结果合成的语音效果并不劣于ＭＢＥ 标准算法     

小波激励线性预测编码（WELP）：一种新的低速语音编码方法

在低速语音编译码系统中，常采用码本激励线性预测编码ＣＥＬＰ，其中随机码本的码本结构及应的索算法直接影响着语音编译码系统的语音质量和实时实现中的运算量。     

语音增强IMBE声码器研究

多带激励声码器（ＭＢＥ）由ＭＩＴ的Ｇｒｉｆｉｎ在１９８７年提出，其改进算法（ＩＭＢＥ）已被ＩＮ－ＭＡＲＳＡＴ采纳作为卫星话音通信的标准。ＭＢＥ声码器在中低速率上可获得较好的合成语音质量，但在噪声环境中使用时，随着输入信噪比的降低，其性能将显著恶化。本文试图将语音增强技术与ＭＢＥ模型相结合以提高声码器抗噪声的性能。我们研究了两种方案：一是采用语音增强预处理器和ＩＭＢＥ声码器级联，二是将语音增强技术和ＩＭＢＥ声码器有机结合构成语音增强ＩＭＢＥ声码器。客观测试和主观试听表明，这两种系统在噪声环境中工作时，性能都有很大的提高。     

一种2.4kbps改进型MBELP编码

该文给出了一种改进的２．４ｋｂ／ｓ多带激励线性预测（ＩＭＢＥＬＰ）语音编码算法,与传统的ＭＢＥＬＰ算法相比,本算法在音质提取和清／浊音判决上采取了一些改进措施,使得合成语音质量有一定的提高。本文详细介绍了改进后的ＭＢＥＬＰ算法,并将其在基音提取和清／浊音判决的结果与传统的ＭＢＥＬＰ进行比较。     

CELP短时预测器算法的研究

码激励线性预测能够在低比特率情况下实现较高质量的语音，在ＣＥＬＰ编码方案的实现中，确定短时预测器和长时预测器的系数是至关重要的，在简单介绍了ＣＥＬＰ的基本原理和激励码本的产生方法后，着重研究短时预测器的算法，采用Ｂｕｒｇ法实现短时预测器，并正对其参数量化编码，计算机模拟表明，采用上述方法可得到较好的合成语音质量。     

1.8kb/s的多带线性预测语音编码算法

设计了一种数码率为１．８ｋｂ／ｓ的多带线性预测（ＭＢＬＰ）语音压缩编码算法。该算法采用基于谐振结构的线性预测分析和对激励信号采用多带处理的方法。试验结果表明，本算法提供了相当于码率为２．４ｋｂ／ｓ美国联邦声码器标准ＭＥＬＰ的重建语音质量，具有较高的清晰度和自然度。     

基于DyWT基音检测的改进MBE语音压缩算法

在对ＭＢＥ算法的复杂度进行深入分析的基础上，本文提出了一种基于二进离散小波变换（ＤｙＷＴ）基音检测法的改进ＭＢＥ语音在压缩算法，并利用仿真结果，对改进前，后的ＭＢＥ算法性能作了一个全面的地比分析，从而得出这样的结论：改进的ＭＢＥ算法保持了原算法各项优点的基础上，大大降低了其复杂度。     

RS码频域编译码的计算机模拟

频域编译码是近年来由Ｂｌａｈｕｔ等人提出来的纠错码编译方法。本文介绍了在时域和频域编译ＲＳ码的基本方法，并给出了在移动信息道的简单分群Ｍａｒｋｏｖ模型下频域编译码的计算机模拟方法和结果。     

卫星通信高速数据传输纠错技术新进展

本文扼要介绍最近卫星通信高速数据传输中采用纠错编译码技术的新进展，说明１２０Ｍｂ／ｓＴＤＭＡ／ＤＳＩ系统中ＢＣＨ（１２８，１１２）码，４５Ｍｂ／ｓＩＤＲ Ｒ３／４Ｖｉｔｅｒｂｉ译码，以及７０Ｍｂ／ｓ卷积码与Ｒｅｅｄ－Ｓｏｌｏｍａｎ码级联等三种典型纠错的生成多项式，并行编译码技术及性能。它们对于卫星通信传输高速数字图像信号是非常有用的。     

采用软判决的不等保护度码

为了使纠错码技术更有效地应用于数字信道，本文提出使用不等保护等码实现不相等保护度的数据传输，并用软判决译码方法生成不等保护度码，还以数字微波通信中的ＰＣＭ信号为例，介绍了其编译过程。     

一种改进的信号灵活分割算法及语音信号清-浊音的自动分割

文中主要对王永忠等提出的灵活分割算法存在的问题做了相应的改进,并做了比较分析,然后将改进后的分割算法应用于语音信号的清-浊音自动分割中.经过大量的理论模型与实际语音信号验证该改进后的算法确实解决了二进分割算法及王永忠方法存在的问题,达到了对信号自适应有效分割.仍然采用Wesfreid等提出的清-浊音识别准则,将新的分割方法应用到实际语音信号的清-浊音自动分割中,不仅同样产生较好划分结果,而且在时间上没有过多的冗余分割.     

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。     

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。     

变码率分类型实时语音声码器

本文提出的声码器将语音分成静音、清音、浊音和混合音四类。用自适应方法进行分频带清浊音判决和有声／无声判决,提高了分类算法的稳定性、准确性和灵活性、准确性和灵活性,还保持了混合语音的音质,且无须对清浊音判决结果进行编码。对清音和浊音的频谱分别采用不同的ＬＳＰ量化表进行编码,从而用标量量化器替代子矢量量化器,降低了复杂度。声码器的码率最高２．４ｋｂｐｓ,最低为１００ｂｐｓ,平均码率１．４ｋｂｐｓ。实时     

Real-time implementation and performance evaluation of speech classifiers in speech analysis-synthesis

In this work, six voiced/unvoiced speech classifiers based on the autocorrelation function (ACF), average magnitude difference function (AMDF), cepstrum, weighted ACF (WACF), zero crossing rate and energy of the signal (ZCR-E), and neural networks (NNs) have been simulated and implemented in real time using the TMS320C6713 DSP starter kit. These speech classifiers have been integrated into a linear-predictive-coding-based speech analysis-synthesis system and their performance has been compared in terms of the percentage of the voiced/unvoiced classification accuracy, speech quality, and computation time. The results of the percentage of the voiced/unvoiced classification accuracy and speech quality show that the NN-based speech classifier performs better than the ACF-, AMDF-, cepstrum-, WACF- and ZCR-E-based speech classifiers for both clean and noisy environments. The computation time results show that the AMDF-based speech classifier is computationally simple, and thus its computation time is less than that of other speech classifiers, while that of the NN-based speech classifier is greater compared with other classifiers.     

Measuring and modeling vocal source-tract interaction

The quality of synthetic speech is affected by two factors: intelligibility and naturalness. At present, synthesized speech may be highly intelligible, but often sounds unnatural. Speech intelligibility depends on the synthesizer's ability to reproduce the formants, the formant bandwidths, and formant transitions, whereas speech naturalness is thought to depend on the excitation waveform characteristics for voiced and unvoiced sounds. Voiced sounds may be generated by a quasiperiodic train of glottal pulses of specified shape exciting the vocal tract filter. It is generally assumed that the glottal source and the vocal tract filter are linearly separable and do not interact. However, this assumption is often not valid, since it has been observed that appreciable source-tract interaction can occur in natural speech. Previous experiments in speech synthesis have demonstrated that the naturalness of synthetic speech does improve when source-tract interaction is simulated in the synthesis process. The purpose of this paper is two-fold: (1) to present an algorithm for automatically measuring source-tract interaction for voiced speech, and (2) to present a simple speech production model that incorporates source-tract interaction into the glottal source model, This glottal source model controls: (1) the skewness of the glottal pulse, and (2) the amount of the first formant ripple superimposed on the glottal pulse. A major application of the results of this paper is the modeling of vocal disorders     

一种新的子波域语音增强方法

提出了一种新的子波域语音增强法,即首先对带噪语音进行1层离散小波变换,然后对提取出来的低频信号和高频信号分别作3层DWT和3层小波包分解,最后对去噪后的语音完成重构。为了在降噪过程中减少清音信息的损失, 文中对语音信号进行了清浊音判决并分别采用多阈值进行处理。计算机仿真结果表明,经本文方法增强语音的清音成分得到了较好保留,并且增强语音的主客观质量均优于DWT去噪法和WPD去噪法。     

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.     

Strategies to improve the performance of very low bit rate speech coders and application to a variable rate 1.2 kb/s codec

This paper presents several strategies to improve the performance of very low bit rate speech coders and describes a speech codec that incorporates these strategies and operates at an average bit rate of 1.2 kb/s. The encoding algorithm is based on several improvements in a mixed multiband excitation (MMBE) linear predictive coding (LPC) structure. A switched-predictive vector quantiser technique that outperforms previously reported schemes is adopted to encode the LSF parameters. Spectral and sound specific low rate models are used in order to achieve high quality speech at low rates. An MMBE approach with three sub-bands is employed to encode voiced frames, while fricatives and stops modelling and synthesis techniques are used for unvoiced frames. This strategy is shown to provide good quality synthesised speech, at a bit rate of only 0.4 kb/s for unvoiced frames. To reduce coding noise and improve decoded speech, spectral envelope restoration combined with noise reduction (SERNR) postfilter is used. The contributions of the techniques described in this paper are separately assessed and then combined in the design of a low bit rate codec that is evaluated against the North American Mixed Excitation Linear Prediction (MELP) coder. The performance assessment is carried out in terms of the spectral distortion of LSF quantisation, mean opinion score (MOS), A/B comparison tests and the ITU-T P.862 perceptual evaluation of speech quality (PESQ) standard. Assessment results show that the improved methods for LSF quantisation, sound specific modelling and synthesis and the new postfiltering approach can significantly outperform previously reported techniques. Further results also indicate that a system combining the proposed improvements and operating at 1.2 kb/s, is comparable (slightly outperforming) a MELP coder operating at 2.4 kb/s. For tandem connection situations, the proposed system is clearly superior to the MELP coder.     

Hardware Realization of Waveform Vector Quantizers

A real-time full search vector quantization system for speech waveform coding is implemented using LSTTL and CMOS devices. The system consists of low-pass filters, A/D and D/A converters, an algorithm for discriminating voiced and unvoiced speed, a full search vector quantizer encoder and decoder, and a microprocessor-based controller. The system is designed to operate at two possible rates: one bit/sample using a dimension 8 vector quantizer (6500 bits/s) or 2 bits/sample using a dimension 4 vector quantizer (13 000 bits/s). In both cases the codebooks have rate 8 bits/vector. Separate codebooks were designed for voiced and unvoiced speech based on a training sequence of 640 000 samples containing five different speakers. The subjective and quantitative results are compared to both simulations and with a real-time array processor based implementation.