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11. 发明申请

WO1991013432A1 DYNAMIC CODEBOOK FOR EFFICIENT SPEECH CODING BASED ON ALGEBRAIC CODES 审中-公开
标题翻译：基于代码的高效语音编码动态代码
公开(公告)号：WO1991013432A1
公开(公告)日：1991-09-05
申请号：PCT/CA1990000381
申请日：1990-11-06
申请人： UNIVERSITE DE SHERBROOKE , ADOUL, Jean-Pierre , LAFLAMME, Claude
发明人： UNIVERSITE DE SHERBROOKE
IPC分类号： G10L09/14
CPC分类号： G10L19/10 , G10L19/00 , G10L19/12 , G10L25/06 , G10L2019/0004 , G10L2019/0008 , G10L2019/0011
摘要： A method of encoding a speech signal is disclosed. This method improves the excitation codebook and search procedure of the conventional Code Excited Linear Prediction (CELP) speech encoders. Use is made of a dynamic codebook (201, 202) based on a combination of two modules: a sparce algebraic code generator (201) associated to a filter (202) having a transfer function varying in time. The generator (201) is a structured codebook with codewords having very few non zero components. The filter (202) shapes the spectral characteristics whereby the resulting excitation codebook (201, 202) exhibits favorable perceptual properties. The search complexity in finding the best codeword is greatly reduced by bringing the search back to the algebraic code domain thereby allowing the sparcity of the algebraic code to speed up the necessary computations.
摘要翻译：公开了一种对语音信号进行编码的方法。该方法改进了常规码激励线性预测（CELP）语音编码器的激励码本和搜索过程。使用基于两个模块的组合的动态码本（201,202）：与具有随时间变化的传递函数的滤波器（202）相关联的码元代数生成器（201）。发生器（201）是具有非常少的非零分量的码字的结构化码本。滤光器（202）形成光谱特性，由此产生的激励码本（201,202）表现出有利的感知特性。通过将搜索引回代数代码域，从而大大减少了找到最佳代码字的搜索复杂度，从而允许代数代码的稀疏性加快了必要的计算。

12. 发明申请

WO2014202539A1 APPARATUS AND METHOD FOR IMPROVED CONCEALMENT OF THE ADAPTIVE CODEBOOK IN ACELP-LIKE CONCEALMENT EMPLOYING IMPROVED PITCH LAG ESTIMATION 审中-公开
标题翻译：用于改进自适应代码的ACARP类似设计的方法和方法，用于改进PITCH LAG估计
公开(公告)号：WO2014202539A1
公开(公告)日：2014-12-24
申请号：PCT/EP2014/062589
申请日：2014-06-16
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V.
发明人： LECOMTE, Jérémie , SCHNABEL, Michael , MARKOVIC, Goran , DIETZ, Martin , NEUGEBAUER, Bernhard
IPC分类号： G10L19/005 , G10L25/90 , G10L19/107 , G10L19/08
CPC分类号： G10L19/005 , G10L19/08 , G10L19/107 , G10L19/125 , G10L25/90 , G10L2019/0002 , G10L2019/0003 , G10L2019/0008
摘要： An apparatus for determining an estimated pitch lag is provided. The apparatus comprises an input interface (1 10) for receiving a plurality of original pitch lag values, and a pitch lag estimator (120) for estimating the estimated pitch lag. The pitch lag estimator (120) is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to said original pitch lag value.
摘要翻译：提供了一种用于确定估计音调滞后的装置。该装置包括用于接收多个原始音调滞后值的输入接口（110）和用于估计估计音调滞后的音调滞后估计器（120）。音调滞后估计器（120）被配置为根据多个原始音调滞后值并根据多个信息值来估计估计音调滞后，其中对于多个原始音调滞后值的每个原始音调滞后值，将多个信息值的信息值分配给所述原始音调滞后值。

13. 发明申请

WO2010047566A2 AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF 审中-公开
标题翻译：一种处理音频信号的装置及其方法
公开(公告)号：WO2010047566A2
公开(公告)日：2010-04-29
申请号：PCT/KR2009/006184
申请日：2009-10-26
申请人： LG ELECTRONICS INC. , YOON, Sung Yong , LEE, Hyun Kook , KIM, Dong Soo , LIM, Jae Hyun
发明人： YOON, Sung Yong , LEE, Hyun Kook , KIM, Dong Soo , LIM, Jae Hyun
IPC分类号： H03M7/30 , G10L19/00
CPC分类号： G10L19/18 , G10L19/0212 , G10L2019/0008
摘要： An apparatus for processing an audio signal and method thereof are disclosed, by which extracting, by an audio processing apparatus, scheme type information indicating either time excitation scheme or frequency excitation scheme for each of a plurality of subframes included in current frame; when the time excitation scheme is applied to at least one subframe among the plurality of subframes according to the scheme type information, extracting mode information representing bit allocation of codebook index for the current frame; and, when the time excitation scheme is applied to at least one subframe among the plurality of subframes according to the scheme type information, decoding the at least one subframe using the mode information according to the time excitation scheme.
摘要翻译：公开了一种用于处理音频信号的设备及其方法，通过该设备，由音频处理设备提取指示多个子帧中的每一个的时间激励方案或频率激励方案的方案类型信息包含在当前框架中; 当根据所述方案类型信息将所述时间激励方案应用于所述多个子帧中的至少一个子帧时，提取表示当前帧的码本索引的比特分配的模式信息; 并且当根据所述方案类型信息将所述时间激励方案应用于所述多个子帧中的至少一个子帧时，根据所述时间激励方案使用所述模式信息对所述至少一个子帧进行解码。

14. 发明申请

WO2010003564A1 LOW BITRATE AUDIO ENCODING/DECODING SCHEME HAVING CASCADED SWITCHES 审中-公开
标题翻译：低成本音频编码/解码方案具有CASCADED开关
公开(公告)号：WO2010003564A1
公开(公告)日：2010-01-14
申请号：PCT/EP2009/004652
申请日：2009-06-26
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V , VOICEAGE CORPORATION , GRILL, Bernhard , LEFEBVRE, Roch , BESSETTE, Bruno , LAPIERRE, Jimmy , GOURNAY, Philippe , SALAMI, Redwan , BAYER, Stefan , FUCHS, Guillaume , GEYERSBERGER, Stefan , GEIGER, Ralf , HILPERT, Johannes , KRAEMER, Ulrich , LECOMTE, Jeremie , MULTRUS, Markus , NEUENDORF, Max , POPP, Harald , RETTELBACH, Nikolaus
发明人： GRILL, Bernhard , LEFEBVRE, Roch , BESSETTE, Bruno , LAPIERRE, Jimmy , GOURNAY, Philippe , SALAMI, Redwan , BAYER, Stefan , FUCHS, Guillaume , GEYERSBERGER, Stefan , GEIGER, Ralf , HILPERT, Johannes , KRAEMER, Ulrich , LECOMTE, Jeremie , MULTRUS, Markus , NEUENDORF, Max , POPP, Harald , RETTELBACH, Nikolaus
IPC分类号： G10L19/14
CPC分类号： G10L19/008 , G10L19/0017 , G10L19/0212 , G10L19/173 , G10L19/18 , G10L2019/0008
摘要： An audio encoder comprises a first information sink oriented encoding branch such as a spectral domain encoding branch, a second information source or SNR oriented encoding branch such as an LPC-domain encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch comprises a converter into a specific domain different from the spectral domain such as an LPC analysis stage generating an excitation signal, and wherein the second encoding branch furthermore comprises a specific domain coding branch such as LPC domain processing branch, and a specific spectral domain coding branch such as LPC spectral domain processing branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder comprises a first domain decoder such as a spectral domain decoding branch, a second domain decoder such as an LPC domain decoding branch for decoding a signal such as an excitation signal in the second domain, and a third domain decoder such as an LPC-spectral decoder branch and two cascaded switches for switching between the decoders.
摘要翻译：音频编码器包括第一信息宿定向编码分支，诸如频域编码分支，第二信息源或SNR定向编码分支，诸如LPC域编码分支，以及用于在第一编码分支与第二编码分支之间进行切换的开关编码分支，其中所述第二编码分支包括转换器到不同于所述频域的特定域，例如产生激励信号的LPC分析阶段，并且其中所述第二编码分支还包括特定域编码分支，诸如LPC域处理分支，以及诸如LPC频域处理分支的特定频域编码分支，以及用于在特定编码分支和特定频域编码分支之间切换的附加开关。音频解码器包括诸如频域解码分支的第一域解码器，诸如LPC域解码分支的第二域解码器，用于对诸如第二域中的激励信号的信号进行解码，以及第三域解码器，诸如LPC 光谱解码器分支和两个级联开关，用于在解码器之间切换。

15. 发明申请

WO2009010831A1 FLEXIBLE PARAMETER UPDATE IN AUDIO/SPEECH CODED SIGNALS 审中-公开
标题翻译：音频/语音编码信号中的灵活参数更新
公开(公告)号：WO2009010831A1
公开(公告)日：2009-01-22
申请号：PCT/IB2007/052866
申请日：2007-07-18
申请人： Nokia Corporation , OJALA, Pasi , LAKANIEMI, Ari
发明人： OJALA, Pasi , LAKANIEMI, Ari
IPC分类号： G10L21/04
CPC分类号： G10L21/04 , G10L19/07 , G10L2019/0008
摘要： This invention relates to a method, a computer program product, apparatuses and a system for extracting coded parameter set from an encoded audio/speech stream, said audio/speech stream being distributed to a sequence of packets, and generating a time scaled encoded audio/speech stream in the parameter coded domain using said extracted coded parameter set.
摘要翻译：本发明涉及一种从编码音频/语音流中提取编码参数集的方法，计算机程序产品，装置和系统，所述音频/语音流被分配到分组序列，并且生成时间缩放编码音频/ 使用所述提取的编码参数集在参数编码域中的语音流。

16. 发明申请

WO2007143604A2 PACKET LOSS CONCEALMENT FOR A CONJUGATE STRUCTURE ALGEBRAIC CODE EXCITED LINEAR PREDICTION DECODER 审中-公开
标题翻译：连接结构代码代码的分组损失隐藏线性预测解码器
公开(公告)号：WO2007143604A2
公开(公告)日：2007-12-13
申请号：PCT/US2007/070319
申请日：2007-06-04
申请人： TEXAS INSTRUMENTS INCORPORATED , LI, Dunling
发明人： LI, Dunling
IPC分类号： G10L11/04
CPC分类号： G10L19/005 , G10L2019/0008
摘要： A method to improve packet loss concealment for generation of a synthetic speech signal in an algebraic code excited linear prediction decoder for a voice over packet network. One method improves features for coding gains in the decoder and for post-filtering of the signals. An alternative method uses a classification method for the signal based on the bitstream in the decoder. In a described example, a method of excitation signal level adjustment (80) after packet loss can be applied to fixed codebook gain (42) of the next good frame through multiplexer (82). During the packet loss, the pulse positions of fixed codebook (28) are unknown, thus it can be difficult to predict them correctly. Wrong pulse locations within a large gain (42) can cause severe distortion on synthesized signals of lost frames and the contiguous good frames in the rest of speech frames. Therefore, zero fixed codebook gain is used in lost frames, which is the standard recommendation in G.729. To composite the fixed codebook contribution, the beginning of the next good frame will adjust the excitation signal level based on the current codebook gain and lost frame duration. The excitation signal level adjustment is applied to adjust the gain error. Further embodiments of improving the PLC strategy in decoder is the excitation signal level adjustment (80) applied to the fixed codebook (28) gain (42).
摘要翻译：一种用于在用于语音分组网络的代数激励线性预测解码器中用于生成合成语音信号的分组丢失隐藏的方法。一种方法改进了用于解码器中的增益编码和用于信号后置滤波的特征。一种替代方法使用基于解码器中的比特流的信号分类方法。在所描述的示例中，分组丢失之后的激励信号电平调整（80）的方法可以通过多路复用器（82）应用于下一个良好帧的固定码本增益（42）。在分组丢失期间，固定码本（28）的脉冲位置是未知的，因此可能难以正确地预测它们。在大增益（42）内的错误的脉冲位置可能导致丢失帧的合成信号和其余语音帧中的连续好帧的严重失真。因此，固定码本增益在零帧中使用，这是G.729中的标准建议。为了复合固定码本贡献，下一个好帧的开始将基于当前码本增益和丢帧持续时间来调整激励信号电平。应用激励信号电平调整来调整增益误差。改进解码器中PLC策略的其它实施例是应用于固定码本（28）增益（42）的激励信号电平调整（80）。

17. 发明申请

WO01024166A1 G.723.1 AUDIO ENCODER 审中-公开
标题翻译： G.723.1音频编码器
公开(公告)号：WO01024166A1
公开(公告)日：2001-04-05
申请号：PCT/SG1999/000096
申请日：1999-09-30
IPC分类号： G10L19/12
CPC分类号： G10L19/12 , G10L2019/0008
摘要： The invention relates to a speech encoder and corresponding encoding method for a dual rate coding system generally in accordance with the ITU-T G.723.1 recommendation, the dual rate coding system using Multi-Pulse Maximum Likelihood Quantization (MP-MLQ) and Algebraic Code-Excited Linear Prediction (ACELP) implementations for respective higher and lower transmission rates, wherein if the higher transmission rate is applicable, the number of gain scale factors searched in quantization steps is reduced, and if the lower transmission rate is applicable, the coding system has a threshold test for entry into the third processing loop of the ACELP implementation. The decoder may be a normal decoder. For example, a G.723.1 decoder may also be used to decode the bit streams encoded by a low complexity G.723.1 encoder in accordance with the invention.
摘要翻译：本发明涉及通常根据ITU-T G.723.1建议的双速率编码系统的语音编码器和相应的编码方法，使用多脉冲最大似然量化（MP-MLQ）的双速率编码系统和代数码用于相应的较高和较低传输速率的线性预测（ACELP）实现，其中如果较高的传输速率可应用，则在量化步骤中搜索的增益比例因子的数量减少，并且如果较低的传输速率可应用，编码系统具有进入ACELP实施的第三个处理循环的阈值测试。解码器可以是正常解码器。例如，G.723.1解码器也可以用于解码根据本发明由低复杂度G.723.1编码器编码的比特流。

18. 发明申请

WO99063523A1 METHOD AND DEVICE FOR VOICE ENCODING 审中-公开
标题翻译：方法和系统语音编码
公开(公告)号：WO99063523A1
公开(公告)日：1999-12-09
申请号：PCT/EP1999/003767
申请日：1999-05-31
IPC分类号： G10L19/00 , G10L19/002 , G10L19/005 , G10L19/10 , G10L19/14 , G10L19/18 , H04L1/00 , G10L9/14
CPC分类号： H04L1/0057 , G10L19/002 , G10L19/005 , G10L19/10 , G10L19/18 , G10L2019/0008
摘要： During a voice encoding, especially according to the CELP principle, voice signals are coded with variable coding rates, whereby the coding rate is altered by a corresponding change in the number of bit positions provided for describing the excitation signals.
摘要翻译：在一个语音编码，尤其是根据所述Codierratenänderung通过在用于激励信号的描述的位位置数的相应变化实现了CELP语音原则信号进行编码用的可变编码率。

19. 发明申请

WO1996021221A1 SPEECH CODING METHOD USING LINEAR PREDICTION AND ALGEBRAIC CODE EXCITATION 审中-公开
标题翻译：使用线性预测和代码代码激励的语音编码方法
公开(公告)号：WO1996021221A1
公开(公告)日：1996-07-11
申请号：PCT/FR1996000017
申请日：1996-01-04
申请人： FRANCE TELECOM , LAMBLIN, Claude
发明人： FRANCE TELECOM
IPC分类号： G10L09/14
CPC分类号： G10L19/107 , G10L2019/0008 , G10L2019/0014
摘要： A method using the algebraic index CELP coding technique. The search for CELP excitation comprises calculating certain components of covariance matrix U = H .H where H is a lower triangular Toeplitz matrix formed from the impulse response of a filter consisting of synthesis filters and a perceptual weighting filter. The stored covariance matrix components are only those of pattern U(posi,p,posi,p) and those of pattern U(posi,p,posj,q), where posi,p and posj,q are the i position and the j position respectively for pulses p and q in the algebraic index codes.
摘要翻译：一种使用代数索引CELP编码技术的方法。搜索CELP激励包括计算协方差矩阵U = H H的某些分量，其中H是由由合成滤波器和感知加权滤波器组成的滤波器的脉冲响应形成的下三角形Toeplitz矩阵。存储的协方差矩阵分量只是模式U（posi，p，posi，p）和模式U（posi，p，posj，q）的组合，其中posi，p和posj，q是i位置，j 位置分别代表代数索引码中的脉冲p和q。

20. 发明申请

WO2017166800A1 丢帧补偿处理方法和装置审中-公开
公开(公告)号：WO2017166800A1
公开(公告)日：2017-10-05
申请号：PCT/CN2016/103481
申请日：2016-10-27
申请人：华为技术有限公司
发明人：刘泽新 , 张兴涛 , 王宾 , 苗磊
IPC分类号： G10L19/005
CPC分类号： G10L19/005 , G10L19/0208 , G10L19/06 , G10L19/083 , G10L19/087 , G10L19/09 , G10L19/107 , G10L25/06 , G10L25/78 , G10L2019/0002 , G10L2019/0008 , G10L2019/0016 , G10L2025/783 , H04L43/08
摘要：一种丢帧补偿处理方法和装置，该方法包括：通过丢失帧标记位确定第i帧是否为丢失帧(101)；当第i帧为丢失帧时，根据第i帧的前N帧的帧间关系和前N帧的帧内关系中的至少一个，估计第i帧的谱频率参数、基音周期和增益(102)。前N帧的帧间关系包括前N帧的相关性和能量稳定性中的至少一个，前N帧的帧内关系包括前N帧中子帧间的相关性和能量稳定性中的至少一个。通过前N帧的信号相关性和能量稳定性，以及每帧帧内信号的相关性和能量稳定性确定第i帧的参数，考虑到了信号之间的关系，使得估计得到的第i帧的参数更加准确，从而提升了解码语音信号的质量。

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