CN101228573A - 利用基音延迟曲线调整对信息信号编码的方法和装置 - Google Patents

利用基音延迟曲线调整对信息信号编码的方法和装置 Download PDF

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CN101228573A
CN101228573A CNA2006800272071A CN200680027207A CN101228573A CN 101228573 A CN101228573 A CN 101228573A CN A2006800272071 A CNA2006800272071 A CN A2006800272071A CN 200680027207 A CN200680027207 A CN 200680027207A CN 101228573 A CN101228573 A CN 101228573A
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pitch delay
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CN101228573B (zh
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詹姆斯·P·阿什利
乌达·米塔尔
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/90Pitch determination of speech signals
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/09Long term prediction, i.e. removing periodical redundancies, e.g. by using adaptive codebook or pitch predictor
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/06Transformation of speech into a non-audible representation, e.g. speech visualisation or speech processing for tactile aids
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility

Abstract

在语音编码/解码器(200/300)中,使用基音延迟曲线端点修正器(208)来将基音延迟插值曲线的端点上下移位。尤其是,根据基音延迟的方差和/或标准偏差来将基音延迟插值曲线的端点移位。

Description

利用基音延迟曲线调整对信息信号编码的方法和装置
技术领域
本发明一般涉及通信系统,尤其是涉及这种通信系统中对信息信号进行编码。
背景技术
数字语音压缩系统通常需要估计输入信号的基频。基频f0通常是根据基音延迟τ0(或被称为“迟滞”)来估计。两者关系由下式表达:
τ 0 = f s f 0 - - - ( 1 )
其中,采样频率fs对于电话级别应用而言通常为8000Hz。
由于语音信号通常为不稳定的,因此被其划分为被称为帧的有限长度矢量,每帧被假定为亚稳态。这些帧的长度通常为10到40毫秒的量级。然后,按照相关的帧长度间隔,对描述该语音信号的参数进行更新。原始的码激励线性预测(CELP)算法按照更短的子帧间隔,进一步更新基音周期信息(利用被称为长期预测的方法,或称为LTP),因此能够实现帧到帧的更平滑过渡。还曾注意到,尽管可以使用开环方法估计τ0,但是使用闭环方法可以得到好得多的性能。闭环方法涉及以子帧为基础,对τ0的不同可能值(通常为从20到147的整数值)进行反复试验搜索,并选择满足一些最低误差标准的值。
这种方法的改进方法包括允许τ0取整数加小数值,如美国专利No.US5359696中给出的那样。这种方法的实际应用的例子能够在GSM半速率语音编码器中找到,如附图1所示,并在美国专利No.US5253269中有说明。这里,21到22-2/3范围内的迟滞是允许的1/3采样分辨率,23到34-5/6范围内的迟滞是允许的1/6采样分辨率,等等。为了保持低的搜索复杂度,使用开环方法和闭环方法的组合。开环方法涉及使用自相关峰值拾取算法生成整数迟滞候选值列表。随后,闭环方法在该整数迟滞候选值附近的可允许迟滞中搜索最佳小数迟滞值。而且,根据与前一子帧之间的差值,对子帧2、3和4的迟滞进行编码。由于存在迟滞参数的高帧内相关,所以这使得能够使用更少的比特对该迟滞信息进行编码。即使如此,GSM HR编解码器仍然每20毫秒使用总共8+(3×4)=20比特来传送该基音周期信息(1.0kbps)。
在降低基音周期信息的位速率的努力中,已开发了一种插值策略,其允许每帧仅对基音信息编码一次(仅仅使用7比特=>350bps),而不是使用通常的子帧解决方案。这项技术被称为松弛码激励线性预测(或者RCELP)技术,并且成为用于码分多址(CDMA)无线电话系统的增强变速率编解码器(EVRC)标准的基础。其基本原理如下。
为以当前帧的终点为中心的分析窗口,估计基音周期。接着生成迟滞(基音延迟)曲线,其由过去帧迟滞到当前帧迟滞的线性插值构成。接着,借助于复杂多相滤波和偏移技术来修正该去线性预测(LP)残差信号,被设计用于将残差波形与所估计的基音延迟曲线相匹配。此残差修正过程的主要原因是解决开环整数迟滞估计过程的精度限制。例如,如果该整数迟滞被估计为32个样本,当实际上真实的迟滞为32.5个样本时,在单个160采样帧中,该残差波形可以与所估计的延迟冲突2.5个样本之多。这会严重降低LTP的性能。该RCELP算法通过在残差波形中的感知上不重要的情形期间(例如,低能量)对残差波形进行移位以与所估计的基音延迟曲线匹配,来解决这个问题。通过修正残差波形以与所估计的基音延迟曲线匹配,保留了LTP的效率,并维持了编码增益。另外,要求可以忽略由于残差修改引起的相关的感知衰退。
美国专利No.6,113,653中提出了一种对处理基音延迟曲线信息的进一步改进,其中公开了一种按照在长度上小于或等于一个块的间隔来调整基音延迟曲线的方法。在此方法中,根据某一误差最小标准,使用少量的比特来对基音延迟曲线的调整进行编码。该方法描述了通过将累积的移位参数最小化,或者将感知加权的输入语音和通过感知加权合成滤波器的自适应码本(ACB)贡献率之间的互相关最大化,来调整基音延迟曲线的技术。其他公知的基音延迟调整标准也可以包括将目标语音和滤波后的ACB贡献率之间的感知加权误差能量最小化。
尽管此方法使用了非常有效的技术来估计和编码基音延迟曲线调整信息,但是该低的位速率对正被编码的基音延迟调整参数的分辨率和/或动态范围产生约束。因此,现在需要通过自适应地修正预测器步长的动态范围和分辨率来改进低位速率长期预测器的性能,以便与现有技术相比,对于给定的位速率获得较高长期预测增益,或者替换地,在较低的位速率上获得相似的长期预测。
附图说明
图1为现有技术的语音编码器的框图。
图2为语音编码器的框图。
图3为语音解码器的框图。
图4图示了在时域中显示的信号的图形表示。
图5为示出图2和图3的编码器和解码器的操作的流程图。
具体实施方式
一般说来,开环基音延迟曲线估计器在对信息信号进行编码期间生成基音延迟信息。以子帧为基础对基音延迟曲线(例如,过去帧迟滞到当前帧迟滞的线性插值)进行调整,这允许对真实的基音延迟曲线作出更精确的估计。基音延迟曲线重建模块在重建帧间的信息信号时,在解码器中使用基音延迟信息。在本发明的优选实施例中,对于基音延迟曲线的调整是基于基音延迟(τ0)中的标准偏差和/或方差。
更具体来说,对信息信号进行编码的方法包括如下步骤:把信息信号划分成块,估计该信息的当前和先前块的基音延迟,并根据τ0中的过去的变化(例如,标准偏差和/或方差)来形成基音延迟的调整。该方法进一步包括如下步骤:按照在长度上小于或等于一个块的间隔来调整基音延迟曲线的形状,对调整后的基音延迟曲线进行编码,以产生适于向目的地传送的代码。
按照在长度上小于或等于一个块的间隔来调整基音延迟曲线的形状的步骤进一步包括如下步骤:确定在位于当前和先前基音延迟处或两者之间的点处的调整后基音延迟,在先前的基音延迟点和调整后的基音延迟点之间形成线性插值。当确定调整后的基音延迟点时,将累积移位的变化最小化。确定调整后的基音延迟的步骤进一步包括如下步骤:将目标残差信号和原始残差信号之间的相关最大化。先前的基音延迟点进一步包括先前的调整后基音延迟点。替代地,对基音延迟曲线的形状进行调整的步骤进一步包括如下步骤:确定位于当前和先前基音延迟处或者两者之间的多个调整后基音延迟点,并在调整后延迟点之间形成线性插值。
还公开了对信息信号进行编码的系统。该系统包括编码器,该编码器包括用于把该信息信号划分成块的装置,以及用于估计信息中的当前和先前块的基音延迟、并根据τ0中的过去的变化(例如,标准偏差和/或方差)来调整基音延迟的装置。
在该系统内,该信息信号进一步包括语音或者音频信号,该信息信号的块进一步包括信息信号的帧。基音延迟信息进一步包括基音延迟调整索引。该系统还包括用于接收基音延迟信息、并产生用于重建该信息信号的调整后基音延迟曲线τc(n)的解码器。
附图2概括描述了依照本发明优选实施例的采用自适应步长基音延迟调整的语音压缩系统200。如附图2所示,通过把输入语音信号s(n)的短期谱包络变平坦的线性预测(LP)分析滤波器202,来处理该输入语音信号s(n)。该LP分析滤波器的输出被指定为LP残差ε(n)。接着,该LP残差ε(n)被开环基音延迟估计器204用来生成该开环基音延迟τ(m)。(此过程的详细内容和下面讨论的一些其他过程在TIA-127EVRC中给出。)接着,该开环基音延迟τ(m)被基音延迟插值块206用来依照下面的表达式生成子帧延迟插值端点矩阵d(m’,j):
d ( m &prime; , j ) = &tau; ( m ) , | &tau; ( m ) - &tau; ( m - 1 ) | > 15 ( 1 - f ( j ) ) &tau; ( m - 1 ) + f ( j ) &tau; ( m ) , otherwise , 1 &le; m &prime; < 3 - - - ( 2 )
其中τ(m)是用于当前帧m的估计开环基音延迟,其以端部的当前帧为中心,τ(m-1)为用于前一帧m-1的估计开环基音延迟,f(n)为一组基音延迟插值系数,其可以被定义为:
         f={0.0,0.3313,0.6625,1.0}             (3)
这些系数例如是当子帧的数量为3(例如,0≤m′≤3)时给出,不过也可以为不是3的子帧值导出一组合适的系数。
同样将开环基音延迟τ(m)用作输入的还有基音延迟变率估计器214。依照当前发明,开环基音延迟估计的采样标准偏差被定义为:
&sigma; &tau; = 1 N - 1 &Sigma; i = 0 N - 1 ( &tau; ( m - i ) - &tau; &OverBar; ) 2 - - - ( 4 )
其中该采样均值
Figure S2006800272071D00053
被定义为:
&tau; &OverBar; = 1 N &Sigma; i = 0 N - 1 &tau; ( m - i ) - - - ( 5 )
当观测数目是二(N=2)时,可以看出,上面的表达式就能够被简化为如下:
&sigma; &tau; = 1 2 | &tau; ( m ) - &tau; ( m - 1 ) | - - - ( 6 )
随后,该变率估计στ和开环基音延迟τ(m)被用作自适应步长生成器215的输入,在该自适应步长生成器215中,作为στ的函数如下计算适应性步长δ(m):
&delta; ( m ) = &alpha; ( &sigma; &tau; ) ( &tau; ( m ) + &tau; ( m - 1 ) 2 ) , - - - ( 7 )
其中α(στ)为该基音延迟的变率估计的某个函数。对于本发明的优选实施例,这个函数如下给出:
      α(στ)=min(Aστ+B,αmax)             (8)
其中A和B可以是常数,στ表示τ的标准偏差,αmax可以是α(στ)的某个最大的允许值。
该自适应步长δ(m)被输入到延迟调整系数生成器216中,在该延迟调整系数生成器216中,可以如下作为基音延迟调整索引i的函数来计算基音延迟调整值Δadj(i):
Δadj(i)=(i-M/2)·δ(m),i∈{0,1,…,M-1}          (9)
其中M为候选基音延迟调整索引的数量。
从上述等式中可以看到,基音延迟调整值Δadj(i)可以取步长δ(m)的整数倍,其中δ(m)不仅仅是基音延迟的平均值(均值)的函数(如现有技术中),而且还是基音延迟值τ(m)的变率估计στ的函数。随后,根据某个失真度量来评估多个基音延迟调整值,并且作为结果,该基音延迟调整值的最佳值可以在编码过程的所有剩余部分中使用。在优选实施例中,该失真度量为第i个滤波后自适应码本贡献率λ(i,n)与加权目标信号sw(n)之间的感知加权均方差。这个过程是在基音延迟调整索引搜索218中给出的,并可以表达为:
i * = arg max i &Element; 0,1 , . . . , M - 1 [ ( &Sigma; n = 0 L - 1 s w ( n ) &lambda; ( i , n ) ) 2 &Sigma; n = 0 L - 1 &lambda; 2 ( i , n ) ] - - - ( 10 )
其中i*是与根据方括弧中的表达式获得的最大值对应的最佳基音延迟调整索引。
为了获得在等式10中使用的信号,使用了基音延迟曲线端点修正器208,以依照下面的表达式将基音延迟插值曲线上下移位:
           d′(m′,j)=d(m′,j)+Δadj(i)            (11)
根据这个表达式,计算210候选基音延迟曲线τc(n),以及获得212自适应码本贡献率E(n)并对其滤波220,以获得滤波后自适应码本贡献率λ(n),如现有技术中那样。
在操作期间,通过发射机200发送诸如固定码本索引、FCB和ACB增益索引等标准变量。连同这些值一起,与用于当前帧τ(m)的基音延迟值的代码一起发送每个子帧的延迟调整索引(i)。来自先前发送帧τ(m-1)中的基音延迟也被使用。该解码器将使用i,τ(m),和τ(m-1)产生相继的基音延迟值之间的插值曲线。更详细来讲,接收器将按照上面讨论的,作为基音延迟调整索引i的函数来计算Δadj(i),并根据等式11应用Δadj(i),以将基音延迟插值曲线的端点向上或向下移位。
附图3为接收器300的框图。如图所示,通过延迟解码器304接收基音延迟参数索引,以产生τ(m)。更具体来讲,解码器304接收表示τ(m)的索引或者“代码”,并对它们进行解码,以产生τ(m)和τ(m-1)。基音延迟值被输出到基音延迟变率估计器214,在该基音延迟变率估计器214处确定基音延迟中的变化,并将其输出到自适应步长生成器215。通过该生成器215计算用于δ(m)的值。该自适应步长被输出到延迟调整系数生成器216中。按照上文所论述的,通过该生成器216作为该基音延迟调整索引i的函数来计算用于Δadj(i)的值,并将其输出到端点修正电路308。
通过发射机200,基音延迟τ(m)被输出到延迟插值块307,并用于根据等式2来生成子帧延迟插值端点矩阵d(m′,j)。延迟曲线端点修正电路308采用该端点矩阵,并依照d′(m′,j)=d(m′,j)+Δadj(i)来将该基音延迟插值曲线的端点上下移位。随后,该移位后的端点被计算电路310用于生成调整后的延迟曲线τc(n),该调整后延迟曲线随后被用于从ACB 312中取得样本(如现有技术)。随后,对ACB贡献率进行调整,并将其与缩放后的固定码本贡献率组合,以生成组合激励信号,该组合激励信号被用作合成滤波器302的输入以产生输出语音信号。该组合激励信号还被用做反馈,以便为下一子帧更新ACB(如现有技术)。
附图4显示如在在时域中显示的先前部分的信号的图形表示。这些信号是基于具有14kHz采样频率的宽带语音编码器结构来采样的。因此,信号402(加权语音信号sw(n))包括一个1/2秒采样(7000个样本)。对于这个例子而言,帧大小为280个样本,子帧大小是70。每个子帧使用一个样本来显示信号404-410。
从该输入信号中估计出开环基音延迟τ(m)404。能够看出,该开环基音延迟估计对于高周期性语音(样本0-2000和4000-6500)而言是相当平滑的,而与之相反,在无声和过渡期间(样本2000-4000和6500-7000)则是相当无规律的。根据本发明,示出了步长δ(m)406。能够看出,当该基音延迟估计的变率小时,该步长也相对小,而相反的,当该基音延迟估计的变率大时,该步长也相对大。可以在最佳基音延迟调整值Δadj(i)408中进一步看出该自适应步长的效果。此处,该最佳基音延迟值仅仅基于4个候选值(每子帧2比特)。在高周期性区域期间,变化很小,并且对分辨率进行强调以允许该基音延迟估计的良好调协。在无声和过渡区域期间,基音延迟变化很大,并随后对宽动态范围进行强调以解决该基音延迟估计中的高不确定性。最后,显示该基音延迟调整后端点d′(m′,1)410,以演示根据本发明的基音延迟曲线的最终综合估计。当与开环基音延迟404相比时,容易看出本发明的整体效果。
附图5是分别示出附图2和附图3中的编码器和解码器的操作的流程图。具体来讲,描述了通过编码器200和解码器300生成基音延迟调整值Δadj。该逻辑流程在步骤501开始,基于输入信号,通过延迟估计电路204或者延迟解码器304来估计基音延迟。在本发明的优选实施例中,该输入信号优选为语音,然而也可以设想其他音频输入信号。在步骤503,基音延迟变率估计器214根据该基音延迟估计来估计基音延迟(τ)的方差和/或标准偏差,以生成自适应步长值δ(m)。更具体来讲,分析τ的过去值以确定στ,δ(m)是按照等式(7),根据στ计算得到的。在步骤505,基音延迟调整系数生成器216使用δ(m),并为调整值(Δadj)确定值。如上面讨论的,Δadj(i)=(i-M/2)·δ(m),i∈{0,1,…,M-1},其中 &delta; ( m ) = &alpha; ( &sigma; &tau; ) ( &tau; ( m ) + &tau; ( m - 1 ) 2 ) . 随后,修正电路208使用用于Δadj的值来生成第二基音延迟参数,并且具体来讲是编码的基音参数(步骤507)。在本发明的优选实施例中,该编码的基音参数包括基音延迟插值曲线的端点,该基音延迟插值曲线是根据该调整值而被向上或者向下移位的,具体来讲是依照该表达式d′(m′,j)=d(m′,j)+Δadj(i),其中i*是与根据等式10获得的最大值对应的最佳基音延迟调整索引。
尽管已参考特定实施例具体示出和说明了本发明,但本领域技术人员可以理解,可以在不背离本发明的精神和范围下,在其中作出形式和细节上的各种变化。例如,尽管在本发明的优选实施例中,基音延迟插值曲线的端点是根据自适应步长而被移位的,但是本领域普通技术人员应认识到可以根据该适应步长生成任一编码的基音参数。更具体来讲,通过允许搜索范围和/或分辨率(即步长)基于基音延迟变率的函数,本发明可以应用到传统的闭环基音延迟和基音搜索方法(例如,美国专利No.5,253,269)。目前这些方法局限于仅仅基于正被搜索的当前基音值的绝对范围的预定分辨率。
在现有解码过程中使用本发明对本领域技术人员来讲也是显而易见的。例如,尽管在本发明的优选实施例中,根据该自适应步长将基音延迟插值曲线的端点向上或者向下移位,但是本领域的普通技术人员将认识到可以根据该自适应步长来生成任一基音延迟参数。如前面讨论的,诸如GSM HR之类的语音解码器可以基于根据任意第一基音延迟参数获得的基音延迟中的变化,使用自适应步长来确定该Δ(delta)编码的迟滞信息(即第二基音延迟参数)的范围和分辨率。因此,该第二基音延迟参数可以基于该自适应步长。
另外,可以使用替代的失真度量,例如累积移位参数的最小化或者归一化互相关参数的最大化(如美国专利No.6,113,653中所说明的),来获得根据本发明的基音延迟曲线调整。对本领域技术人员来讲是显而易见的是:本发明独立于所采用的失真度量,并且可以在不脱离本发明的精神和范围的情况下使用任一方法。

Claims (20)

1.一种操作语音编码器的方法,该方法包括步骤:
根据输入信号估计基音延迟;
根据该基音延迟估计来估计基音延迟中的变化;
根据该基音延迟中的变化来确定自适应步长值;和
根据该自适应步长来生成编码的基音参数。
2.根据权利要求1的方法,其中所述根据输入信号估计基音延迟的步骤包括根据语音或者音频信号来估计该基音延迟的步骤。
3.根据权利要求1的方法,其中所述估计基音延迟中的变化的步骤包括估计基音延迟的方差和/或标准偏差的步骤。
4.根据权利要求1的方法,其中所述确定自适应步长的步骤包括确定自适应步长δ(m)的步骤,其中δ(m)能够被表达为:
&delta; ( m ) = &alpha; ( &sigma; &tau; ) ( &tau; ( m ) + &tau; ( m - 1 ) 2 )
以及其中α(στ)为基音延迟的变率估计的某一函数,以及τ(m)是对于帧号m的基音延迟估计。
5.根据权利要求4的方法,其中α(στ)=min(Aστ+B,αmax),其中A和B为预定值,στ表示τ的标准偏差,以及αmax是α(στ)的最大允许值。
6.根据权利要求1的方法,其中所述根据该自适应步长生成编码的基音参数的步骤包括确定延迟调整值Δadj的步骤,其中
Δadj(i)=(i-M/2)·δ(m),i∈{0,1,…,M-1}
其中M为候选基音延迟调整索引的数量,δ(m)为是自适应步长,以及i∈{0,1,…,M-1}是编码的基音参数。
7.根据权利要求6的方法,其中延迟调整值Δadj被用于根据下面的表达式将基音延迟插值曲线的端点上下移位:
d′(m′,j)=d(m′,j)+Δadj(i)
其中d(m′,j)为子帧延迟插值端点矩阵。
8.根据权利要求1的方法,其中所述根据该自适应步长生成编码的基音参数的步骤包括评估失真标准的步骤。
9.根据权利要求8的方法,其中所述评估失真标准的步骤包括对将均方差参数最小化、将累积的移位参数最小化、以及将归一化的互相关参数最大化的其中之一进行评估的步骤。
10.一种操作语音解码器的方法,该方法包括步骤:
接收第一基音延迟参数;
根据该第一基音延迟参数来估计基音延迟中的变化;
根据该基音延迟中的变化来确定自适应步长;和
根据该自适应步长来生成第二基音延迟参数。
11.根据权利要求10的方法,其中所述估计基音延迟中的变化的步骤包括估计基音延迟的方差和/或标准偏差的步骤。
12.根据权利要求10的方法,其中所述确定自适应步长的步骤包括确定自适应步长δ(m),其中δ(m)可被表达为:
&delta; ( m ) = &alpha; ( &sigma; &tau; ) ( &tau; ( m ) + &tau; ( m - 1 ) 2 )
其中α(στ)是基音延迟的变率估计的某一函数,以及τ(m)是对于帧号m的基音延迟估计。
13.根据权利要求12的方法,其中α(στ)=min(Aστ+B,αmax),其中A和B是预定的,στ表示τ的标准偏差,αmax为α(στ)的最大允许值。
14.根据权利要求10的方法,其中所述根据该自适应步长生成第二基音延迟参数的步骤包括确定延迟调整值Δadj的步骤,其中
Δadj(i)=(i-M/2)·δ(m),i∈{0,1,…,M-1}
其中M为候选基音延迟调整索引的数量,δ(m)为自适应步长。
15.根据权利要求14的方法,其中延迟调整值Δadj被用来根据下面的表达式将基音延迟插值曲线的端点向上或者向下移位:
d′(m′,j)=d(m′,j)+Δadj(i)
其中d(m′,j)为子帧延迟插值端点矩阵,d′(m′,j)为第二基音延迟参数。
16.一种装置,包括:
用于估计基音延迟中的变化的变率估计器;
用于根据该基音延迟中的变化确定自适应步长的系数生成器;和
用于根据该自适应步长修正基音参数的修正电路。
17.根据权利要求16的装置,其中该修正电路根据自适应步长向上或者向下修正基音延迟插值曲线的端点。
18.根据权利要求16的装置,其中该基音延迟基于语音或者音频信号。
19.根据权利要求16的装置,其中基音延迟中变化包括基音延迟的方差和/或标准偏差。
20.权利要求16的装置,其中该自适应步长是按照 &delta; ( m ) = &alpha; ( &sigma; &tau; ) ( &tau; ( m ) + &tau; ( m - 1 ) 2 ) 来计算的,α(στ)是基音延迟的变率估计的某个函数。
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