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1. 发明申请

WO2010006717A1 AUDIO ENCODING/DECODING SCHEME HAVING A SWITCHABLE BYPASS 审中-公开
标题翻译：具有可切换旁路的音频编码/解码方案
公开(公告)号：WO2010006717A1
公开(公告)日：2010-01-21
申请号：PCT/EP2009/004875
申请日：2009-07-06
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , VOICEAGE CORPORATION , GRILL, Bernhard , BAYER, Stefan , FUCHS, Guillaume , GEYERSBERGER, Stefan , GEIGER, Ralf , HILPERT, Johannes , KRAEMER, Ulrich , LECOMTE, Jeremie , MULTRUS, Markus , NEUENDORF, Max , POPP, Harald , RETTELBACH, Nikolaus , LEFEBVRE, Roch , BESSETTE, Bruno , LAPIERRE, Jimmy , SALAMI, Redwan , GOURNAY, Philippe
发明人： GRILL, Bernhard , BAYER, Stefan , FUCHS, Guillaume , GEYERSBERGER, Stefan , GEIGER, Ralf , HILPERT, Johannes , KRAEMER, Ulrich , LECOMTE, Jeremie , MULTRUS, Markus , NEUENDORF, Max , POPP, Harald , RETTELBACH, Nikolaus , LEFEBVRE, Roch , BESSETTE, Bruno , LAPIERRE, Jimmy , SALAMI, Redwan , GOURNAY, Philippe
IPC分类号： G10L19/14
CPC分类号： G10L19/173 , G10L19/0017 , G10L19/008 , G10L19/0212 , G10L19/18 , G10L2019/0008
摘要： An apparatus for encoding comprises a first domain Converter (510), a switchable bypass (50), a second domain Converter (410), a first processor (420) and a second processor (520) to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain Converter allow the generation of a decoded audio signal with high quality and low bit rate.
摘要翻译：一种用于编码的装置，包括第一域转换器（510），可切换旁路（50），第二域转换器（410），第一处理器（420）和第二处理器（520），以获得具有不同信号的编码音频信号由不同域中的编码数据表示的部分，其已经由不同的编码算法编码。解码器中的对应的解码级以及绕过域转换器的旁路允许产生具有高质量和低比特率的解码音频信号。

2. 发明申请

WO02023532A3 SYSTEM OF DYNAMIC PULSE POSITION TRACKS FOR PULSE-LIKE EXCITATION IN SPEECH CODING 审中-公开
标题翻译：语音编码中脉冲激励的动态脉冲位置跟踪系统
公开(公告)号：WO02023532A3
公开(公告)日：2002-05-16
申请号：PCT/IB2001/001731
申请日：2001-09-17
IPC分类号： G10L19/00 , G10L19/10 , G10L21/02
CPC分类号： G10L19/10 , G10L21/0364 , G10L2019/0004 , G10L2019/0008
摘要： A system is disclosed for improving the quality of coded speech information in a communications system. The system dynamically determines pulse tracks that represent an excitation signal. A track or set of tracks that define possible pulse positions are determined based on available information sent to a decoder. Alternatively, at least one first track may include fixed pulse positions, and the remaining tracks may include dynamic pulse positions arranged according to the position of a coded pulse in the first track. Also, all tracks may include dynamically arranged pulse positions that are arranged according to a reference position that is likely to produce a high magnitude pulse signal.
摘要翻译：本发明涉及一种用于改善通信系统中编码的语音信息质量的系统。该系统动态确定代表激励信号的脉冲轨迹。基于发送给解码器的可用信息来确定定义可能的脉冲位置的轨道或一系列轨道。在另一个实施例中，至少一个轨道可以包括固定的脉冲位置，其余轨道可以包括根据第一轨道中的编码脉冲的位置布置的动态脉冲位置。轨道还可以包括根据能够产生高幅度脉冲信号的参考位置布置的动态布置的脉冲位置。

3. 发明申请

WO00054258A1 SOUND SOURCE VECTOR GENERATOR AND VOICE ENCODER/DECODER 审中-公开
标题翻译：声源矢量发生器和语音编码器/解码器
公开(公告)号：WO00054258A1
公开(公告)日：2000-09-14
申请号：PCT/JP2000/001225
申请日：2000-03-02
IPC分类号： G10L19/12 , G10L19/00 , G10L19/08 , H03M7/30 , G10L19/10
CPC分类号： G10L19/107 , G10L19/10 , G10L2019/0008
摘要： A limitation is imposed on a noise code vector generated from an algebra code book so as to reduce the total number of entries of the algebra code book. Entries of a random code book of a large number of pulses are assigned to the reduced part. According to the mode, the number of entries of the reduced part is adaptively changed.
摘要翻译：对从代数码本生成的噪声编码矢量进行限制，以减少代数码本的总数。将大量脉冲的随机码本的条目分配给缩小部分。根据该模式，减少部分的条目数量自适应地改变。

4. 发明申请

WO2010003617A1 LOW BITRATE AUDIO ENCODING/DECODING SCHEME WITH COMMON PREPROCESSING 审中-公开
标题翻译：低分辨率音频编码/解码方案与通用预处理
公开(公告)号：WO2010003617A1
公开(公告)日：2010-01-14
申请号：PCT/EP2009/004873
申请日：2009-07-06
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , GRILL, Bernhard , BAYER, Stefan , FUCHS, Guillaume , GEYERSBERGER, Stefan , GEIGER, Ralf , HILPERT, Johannes , KRAEMER, Ulrich , LECOMTE, Jeremie , MULTRUS, Markus , NEUENDORF, Max , POPP, Harald , RETTELBACH, Nikolaus , NAGEL, Frederik , DISCH, Sascha , HERRE, Juergen , YOKOTANI, Yoshikazu , WABNIK, Stefan , SCHULLER, Gerald , HIRSCHFELD, Jens
发明人： GRILL, Bernhard , BAYER, Stefan , FUCHS, Guillaume , GEYERSBERGER, Stefan , GEIGER, Ralf , HILPERT, Johannes , KRAEMER, Ulrich , LECOMTE, Jeremie , MULTRUS, Markus , NEUENDORF, Max , POPP, Harald , RETTELBACH, Nikolaus , NAGEL, Frederik , DISCH, Sascha , HERRE, Juergen , YOKOTANI, Yoshikazu , WABNIK, Stefan , SCHULLER, Gerald , HIRSCHFELD, Jens
IPC分类号： G10L19/14
CPC分类号： G10L19/173 , G10L19/0017 , G10L19/008 , G10L19/0212 , G10L19/18 , G10L2019/0008
摘要： An audio encoder comprises a common preprocessing stage (100), an information sink based encoding branch (400) such as spectral domain encoding branch, a information source based encoding branch (500) such as an LPC-domain encoding branch and a switch (200) for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage (300). An audio decoder comprises a spectral domain decoding branch, an LPC- domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.
摘要翻译：音频编码器包括公共预处理级（100），诸如频域编码分支的基于信息集合的编码分支（400），诸如LPC域编码分支的信息源编码分支（500）和开关（200 ），用于在由决策阶段（300）控制的这些分支的这些分支或输出的输入处在这些分支之间切换。音频解码器包括频域解码分支，LPC域解码分支，用于在分支之间切换的一个或多个开关和用于后处理时域音频信号以用于获得后处理音频的公共后处理级信号。

5. 发明申请

WO99063522A1 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：WO99063522A1
公开(公告)日：1999-12-09
申请号：PCT/EP1999/003766
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： According to the invention, positions of certain predetermined vector elements which differ from zero, and the signs of these vector elements are coded separately of one another in order to code vectors for describing voice parameters.
摘要翻译：用于语音的描述矢量的编码参数某一预定的非零向量元素和这些向量元素的迹象的位置被彼此分开地编码。

6. 发明申请

WO1996028810A1 DEPTH-FIRST ALGEBRAIC-CODEBOOK SEARCH FOR FAST CODING OF SPEECH 审中-公开
标题翻译：深度第一代码 - 代码搜索快速编码的语音
公开(公告)号：WO1996028810A1
公开(公告)日：1996-09-19
申请号：PCT/CA1996000135
申请日：1996-03-05
申请人： UNIVERSITE DE SHERBROOKE
发明人： UNIVERSITE DE SHERBROOKE , ADOUL, Jean-Pierre , LAFLAMME, Claude
IPC分类号： G10L09/14
CPC分类号： G10L19/12 , G10L19/00 , G10L19/10 , G10L25/06 , G10L2019/0004 , G10L2019/0008 , G10L2019/0011 , G10L2019/0013 , G10L2019/0014
摘要： A codebook is searched in view of encoding a sound signal. This codebook consists of a set of codevectors each of 40 positions and comprising N non-zero-amplitude pulses assignable to predetermined valid positions. To reduce the search complexity, a depth-first search is used which involves a tree structure with levels ordered from 1 through M. A path-building operation takes place at each level whereby a candidate path from the previous level is extended by choosing a predetermined number of new pulses and selecting valid positions for said new pulses in accordance with a given pulse-order rule and a given selection criterion. A path originated at the first level and extended by the path-building operations of subsequent levels determines the respective positions of the N non-zero-amplitude pulse of a candidate codevector. Use of a signal-based pulse-position likelihood estimate during the first few levels enable initial pulse-screening to start the search on favorable conditions. A selection criterion based on maximizing a ratio is used to assess the progress and to choose the best one among competing candidate codevectors.
摘要翻译：考虑到声音信号的编码来搜索码本。该码本由一组40个位置的码矢组成，包括可分配给预定有效位置的N个非零幅度脉冲。为了降低搜索复杂度，使用深度优先搜索，涉及具有从1到M排列的级别的树结构。在每个级别进行路径建立操作，从而通过选择预定的根据给定的脉冲顺序规则和给定的选择标准，为新的脉冲数选择所述新脉冲的有效位置。起始于第一级并由后续级别的路径构建操作扩展的路径确定候选码矢量的N个非零幅度脉冲的相应位置。在前几个级别中使用基于信号的脉冲位置似然估计可使初始脉冲筛选在有利条件下开始搜索。基于最大化比例的选择标准用于评估进度并在竞争的候选代码中选择最佳的选择标准。

7. 发明申请

WO2011048117A1 AUDIO SIGNAL ENCODER, AUDIO SIGNAL DECODER, METHOD FOR ENCODING OR DECODING AN AUDIO SIGNAL USING AN ALIASING-CANCELLATION 审中-公开
标题翻译：音频信号编码器，音频信号解码器，使用消息取消来编码或解码音频信号的方法
公开(公告)号：WO2011048117A1
公开(公告)日：2011-04-28
申请号：PCT/EP2010/065752
申请日：2010-10-19
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , VOICEAGE CORPORATION , KONINKLIJKE PHILIPS ELECTRONICS N.V. , DOLBY INTERNATIONAL AB , BESSETTE, Bruno , NEUENDORF, Max , GEIGER, Ralf , GOURNAY, Philippe , LEFEBVRE, Roch , GRILL, Bernhard , LECOMTE, Jeremie , BAYER, Stefan , RETTELBACH, Nikolaus , VILLEMOES, Lars , SALAMI, Redwan , BRINKER, Albertus C. den
发明人： BESSETTE, Bruno , NEUENDORF, Max , GEIGER, Ralf , GOURNAY, Philippe , LEFEBVRE, Roch , GRILL, Bernhard , LECOMTE, Jeremie , BAYER, Stefan , RETTELBACH, Nikolaus , VILLEMOES, Lars , SALAMI, Redwan , BRINKER, Albertus C. den
IPC分类号： G10L19/14 , G10L19/02
CPC分类号： G10L19/03 , G10L19/0212 , G10L19/20 , G10L2019/0008
摘要： An audio signal decoder (200) for providing a decoded representation (212) of an audio content on the basis of an encoded representation (310) of the audio content comprises a transform domain path (230, 240, 242, 250, 260) configured to obtain a time-domain representation (212) of a portion of the audio content encoded in a transform-domain mode on the basis of a first set (220) of spectral coefficients, a representation (224) of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters (222). The transform domain path comprises a spectrum processor (230) configured to apply a spectrum shaping to the first set of spectral coefficients in dependence on at least a subset of the linear-prediction-domain parameters, to obtain a spectrally-shaped version (232) of the first set of spectral coefficients. The transform domain path comprises a first frequency-domain-to-time-domain converter (240) configured to obtain a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path comprises an aliasing-cancellation stimulus filter configured to filter (250) the aliasing-cancellation stimulus signal (324) in dependence on at least a subset of the linear-prediction-domain parameters (222), to derive an aliasing-cancellation synthesis signal (252) from the aliasing-cancellation stimulus signal. The transform domain path also comprises a combiner (260) configured to combine the time-domain representation (242) of the audio content with the aliasing-cancellation synthesis signal (252), or a post-processed version thereof, to obtain an aliasing reduced time-domain signal.
摘要翻译：用于基于音频内容的编码表示（310）提供音频内容的解码表示（212）的音频信号解码器（200）包括：配置的变换域路径（230,240,242,250,260）基于频谱系数的第一组（220）获得以变换域模式编码的音频内容的一部分的时域表示（212），混叠取消激励信号的表示（224）和多个线性预测域参数（222）。变换域路径包括频谱处理器（230），其被配置为根据线性预测域参数的至少一个子集对第一组频谱系数应用频谱整形，以获得频谱形状的版本（232）的第一组光谱系数。变换域路径包括被配置为基于第一组频谱系数的频谱形式的版本来获得音频内容的时域表示的第一频域 - 时域转换器（240）。变换域路径包括被设置为根据线性预测域参数（222）的至少一个子集过滤（250）混叠取消激励信号（324）的混叠消除刺激滤波器，以导出混叠消除刺激滤波器消除合成信号（252）。变换域路径还包括被配置为将音频内容的时域表示（242）与混叠消除合成信号（252）组合的组合器（260）或其后处理版本，以获得减少的混叠时域信号。

8. 发明申请

WO01020595A1 VOICE ENCODER/DECODER 审中-公开
标题翻译：语音编码器/解码器
公开(公告)号：WO01020595A1
公开(公告)日：2001-03-22
申请号：PCT/JP1999/004991
申请日：1999-09-14
IPC分类号： G10L19/00 , G10L19/04 , G10L19/08 , G10L19/10 , G10L19/14 , G10L9/14
CPC分类号： G10L19/09 , G10L19/04 , G10L19/107 , G10L19/18 , G10L2019/0008
摘要： A voice encoding method using a synthesizing filter having a linear prediction coefficient obtained by dividing an input signal into frames of a fixed length and by linearly predicting/analyzing the input signal in units of a frame and comprising driving the synthesizing filter by a periodic signal outputted from an adaptive code book and a pulsating signal outputted from an algebraic structure code book, producing a reproduced signal, and minimizing the difference between the input signal and the reproduced signal to encode a voice, wherein there are prepared an encoding mode 1 using a pitch lag determined from the input signal of the present frame and an encoding mode 2 using a pitch lag determined from the input signal of a past frame. A mode capable of encoding the input signal more precisely with the encoding mode 1 and the encoding mode 2 is determined for each frame so that the encoding is carried out on the basis of the determined mode.
摘要翻译：一种语音编码方法，使用具有通过将输入信号划分为固定长度的帧而获得的线性预测系数的合成滤波器，并且以帧为单位线性预测/分析输入信号，并且包括通过输出的周期信号驱动合成滤波器从自适应码本和从代数结构码本输出的脉动信号，产生再生信号，并使输入信号与再现信号之间的差异最小化以对语音进行编码，其中准备了使用音调的编码模式1 从当前帧的输入信号确定的滞后和使用从过去帧的输入信号确定的音调滞后的编码模式2。针对每个帧确定能够以编码模式1和编码模式2更准确地对输入信号进行编码的模式，从而基于所确定的模式进行编码。

9. 发明申请

WO99045532A1 SPEECH CODING INCLUDING SOFT ADAPTABILITY FEATURE 审中-公开
标题翻译：语音编码，包括软适应性特征
公开(公告)号：WO99045532A1
公开(公告)日：1999-09-10
申请号：PCT/SE1999/000302
申请日：1999-03-02
IPC分类号： G10L19/002 , G10L19/12 , G10L19/18 , G10L9/14
CPC分类号： G10L19/12 , G10L19/002 , G10L19/18 , G10L2019/0008
摘要： Adaptive speech coding includes receiving an original speech signal, performing on the original speech signal a current coding operation, and adapting the current coding operation (11) in response to information used in the current coding operation (17, 18, 19). Adaptive speech decoding includes receiving coded information, performing a current decoding operation (200) on the coded information, and adapting the current decoding operation in response to information used in the current decoding operation (17, 18, 19).
摘要翻译：自适应语音编码包括接收原始语音信号，对原始语音信号执行当前编码操作，并且响应于当前编码操作（17,18,19）中使用的信息调整当前编码操作（11）。自适应语音解码包括接收编码信息，对编码信息执行当前解码操作（200），并根据当前解码操作（17,18,19）中使用的信息调整当前解码操作。

10. 发明申请

WO99012156A1 REDUCING SPARSENESS IN CODED SPEECH SIGNALS 审中-公开
标题翻译：减少编码语音信号中的空间
公开(公告)号：WO99012156A1
公开(公告)日：1999-03-11
申请号：PCT/SE1998/001515
申请日：1998-08-25
IPC分类号： G10L11/00 , G10L19/00 , G10L19/08 , G10L21/02 , G10L9/14
CPC分类号： G10L19/12 , G10L19/002 , G10L2019/0008
摘要： Sparseness is reduced in an input digital signal (A) which includes a first sequence of sample values. An output digital signal (B) is produced in response to the input digital signal. The output digital signal includes a second sequence of sample values, which second sequence of sample values has a greater density of non-zero sample values than the first sequence of sample values.
摘要翻译：在包括第一序列样本值的输入数字信号（A）中减少了稀疏度。输出数字信号（B）是响应输入的数字信号产生的。输出数字信号包括样本值的第二序列，该样本值的第二序列具有比第一序列样本值更大的非零样本值密度。

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