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1. 发明申请

WO2003102923A3 METHODE AND DEVICE FOR PITCH ENHANCEMENT OF DECODED SPEECH 审中-公开
公开(公告)号：WO2003102923A3
公开(公告)日：2003-12-11
申请号：PCT/CA2003/000828
申请日：2003-05-30
申请人： VOICEAGE CORPORATION , BESSETTE, Bruno , LAFLAMME, Claude , JELINEK, Milan , LEFEBVRE, Roch
发明人： BESSETTE, Bruno , LAFLAMME, Claude , JELINEK, Milan , LEFEBVRE, Roch
IPC分类号： G10L21/02
摘要： In a method and device for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal, the decoded sound signal is divided into a plurality of frequency sub-band signals, and post-processing is applied to at least one of the frequency sub-band signal. After post-processing of this at least one frequency sub-band signal, the frequency sub-band signals may be added to produce an output post-processed decoded sound signal. In this manner, the post-processing can be localized to a desired sub-band or sub-bands with leaving other sub-bands virtually unaltered.

2. 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。

3. 发明申请

WO2003102923A2 A METHOD AND DEVICE FOR FREQUENCY-SELECTIVE PITCH ENHANCEMENT OF SYNTHESIZED SPEECH 审中-公开
标题翻译：用于合成语音的频率选择性增强的方法和设备
公开(公告)号：WO2003102923A2
公开(公告)日：2003-12-11
申请号：PCT/CA2003/000828
申请日：2003-05-30
申请人： VOICEAGE CORPORATION , BESSETTE, Bruno , LAFLAMME, Claude , JELINEK, Milan , LEFEBVRE, Roch
发明人： BESSETTE, Bruno , LAFLAMME, Claude , JELINEK, Milan , LEFEBVRE, Roch
IPC分类号： G10L21/02
CPC分类号： G10L21/0364 , G10L19/26 , G10L21/0232
摘要： In a method and device for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal, the decoded sound signal is divided into a plurality of frequency sub-band signals, and post-processing is applied to at least one of the frequency sub-band signal. After post-processing of this at least one frequency sub-band signal, the frequency sub-band signals may be added to produce an output post-processed decoded sound signal. In this manner, the post-processing can be localized to a desired sub-band or sub-bands with leaving other sub-bands virtually unaltered.
摘要翻译：考虑到提高该解码声音信号的感知质量，对解码声音信号进行后处理的方法和装置中，解码声音信号被分成多个频率子带信号，后处理应用于至少一个频率子带信号。在对该至少一个频率子带信号进行后处理之后，可以添加频率子带信号以产生输出的后处理解码声音信号。以这种方式，后处理可以被定位到期望的子带或子带，而使其他子带几乎不变。

4. 发明申请

WO2012045744A1 APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL AND FOR PROVIDING A HIGHER TEMPORAL GRANULARITY FOR A COMBINED UNIFIED SPEECH AND AUDIO CODEC (USAC) 审中-公开
标题翻译：用于处理音频信号并为组合的统一语音和音频编解码器（USAC）提供较高时间格数的装置和方法
公开(公告)号：WO2012045744A1
公开(公告)日：2012-04-12
申请号：PCT/EP2011/067318
申请日：2011-10-04
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , VOICEAGE CORPORATION , MULTRUS, Markus , GRILL, Bernhard , NEUENDORF, Max , RETTELBACH, Nikolaus , FUCHS, Guillaume , GOURNAY, Philippe , LEFEBVRE, Roch , BESSETTE, Bruno
发明人： MULTRUS, Markus , GRILL, Bernhard , NEUENDORF, Max , RETTELBACH, Nikolaus , FUCHS, Guillaume , GOURNAY, Philippe , LEFEBVRE, Roch , BESSETTE, Bruno
IPC分类号： G10L19/00 , G10L19/02 , G10L21/04
CPC分类号： G10L19/12 , G10L19/0204 , G10L21/00 , G10L2019/0012
摘要： An apparatus for processing an audio signal is provided. The apparatus comprises a signal processor (110; 205; 405) and a configurator (120; 208; 408). The signal processor (110; 205; 405) is adapted to receive a first audio signal frame having a first configurable number of samples of the audio signal, Moreover, the signal processor (110; 205; 405) is adapted to upsample the audio signal by a configurable upsampling factor to obtain a processed audio signal. Furthermore, the signal processor (110; 205; 405) is adapted to output a second audio signal frame having a second configurable number of samples of the processed audio signal. The configurator 120; 208; 408) is adapted to configure the signal processor (110; 205; 405) based on configuration information such that the configurable upsampling factor is equal to a first upsampling value when a first ratio of the second configurable number of samples to the first configurable number of samples has a first ratio value. Moreover, the configurator ( 120; 208; 408) is adapted to configure the signal processor (110; 205; 405) such that the configurable upsampling factor is equal to a different second upsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples has a different second ratio value. The first or the second ratio value is not an integer value.
摘要翻译：提供一种用于处理音频信号的装置。该装置包括信号处理器（110; 205; 405）和配置器（120; 208; 408）。信号处理器（110; 205; 405）适于接收具有音频信号的第一可配置数目的采样的第一音频信号帧。此外，信号处理器（110; 205; 405）适于对音频信号通过可配置的上采样因子来获得经处理的音频信号。此外，信号处理器（110; 205; 405）适于输出具有经处理的音频信号的第二可配置数目的样本的第二音频信号帧。配置器120; 208; 408）适于基于配置信息来配置信号处理器（110; 205; 405），使得当第二可配置数量的采样的第一比率与第一可配置数量的采样值相对应时，可配置上采样因子等于第一上采样值样品具有第一比值。此外，配置器（120; 208; 408）适于配置信号处理器（110; 205; 405），使得可配置上采样因子等于不同的第二上采样值，当第二配置数量的第一可配置数量的样本具有不同的第二比值。第一或第二比值不是整数值。

5. 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。

6. 发明申请

WO2008151410A1 DEVICE AND METHOD FOR NOISE SHAPING IN A MULTILAYER EMBEDDED CODEC INTEROPERABLE WITH THE ITU-T G.711 STANDARD 审中-公开
标题翻译：与ITU-T G.711标准相互配合的多层嵌入式编解码器中的噪声形成装置和方法
公开(公告)号：WO2008151410A1
公开(公告)日：2008-12-18
申请号：PCT/CA2007/002373
申请日：2007-12-28
申请人： VOICEAGE CORPORATION , BESSETTE, Bruno , LAPIERRE, Jimmy , MALENOVSKY, Vladimir , LEFEBVRE, Roch , SALAMI, Redwan
发明人： BESSETTE, Bruno , LAPIERRE, Jimmy , MALENOVSKY, Vladimir , LEFEBVRE, Roch , SALAMI, Redwan
IPC分类号： G10L19/00 , G10L21/02
CPC分类号： G10L19/26 , G10L19/005 , G10L19/24 , G10L25/93
摘要： A device and method for shaping noise during encoding of an input sound signal comprise pre-emphasizing the input signal or a decoded signal from a given sound signal codec to produce a pre-emphasized signal, computing a filter transfer function based on the pre-emphasized signal, and shaping the noise by filtering the noise through the transfer function to produce a shaped noise signal, wherein the noise shaping comprises producing a noise feedback. A device and method for noise shaping in a multilayer codec, including at least Layer 1 and 2, comprise: at an encoder, producing an encoded sound signal in Layer 1 including Layer 1 noise shaping, and producing a Layer 2 enhancement signal; at a decoder, decoding the Layer 1 encoded sound signal to produce a synthesis signal, decoding the enhancement signal, computing a filter transfer function based on the synthesis signal, filtering the enhancement signal through the transfer function to produce a Layer 2 filtered enhancement signal, and adding the filtered enhancement signal to the synthesis signal to produce an output signal including contributions from Layer 1 and 2.
摘要翻译：用于在编码输入声音信号期间整形噪声的装置和方法包括预先强调来自给定声音信号编解码器的输入信号或解码信号以产生预加重信号，基于预先强调的信号计算滤波器传递函数信号和整形噪声，通过传递函数对噪声进行滤波以产生成形噪声信号，其中噪声整形包括产生噪声反馈。包括至少第1层和第2层的多层编解码器中的噪声整形的装置和方法包括：在编码器处，产生包括层1噪声整形的层1中的编码声音信号，并产生第2层增强信号; 在解码器处，解码第1层编码声音信号以产生合成信号，对增强信号进行解码，基于合成信号计算滤波器传递函数，通过传递函数对增强信号进行滤波，以产生第2层滤波的增强信号，并将经滤波的增强信号加到合成信号上以产生包括来自层1和2的贡献的输出信号。

7. 发明申请

WO1998006090A1 SPEECH/AUDIO CODING WITH NON-LINEAR SPECTRAL-AMPLITUDE TRANSFORMATION 审中-公开
标题翻译：语音/音频编码与非线性频谱放大转换
公开(公告)号：WO1998006090A1
公开(公告)日：1998-02-12
申请号：PCT/CA1997000543
申请日：1997-07-30
申请人： UNIVERSITE DE SHERBROOKE , LEFEBVRE, Roch , LAFLAMME, Claude , ADOUL, Jean-Pierre
发明人： UNIVERSITE DE SHERBROOKE
IPC分类号： G10L03/02
CPC分类号： G10L19/0204 , G10L21/0264 , G10L25/27
摘要： In a method and device for encoding a speech and/or audio signal in view of enhancing perceptual quality, the time-domain speech and/or audio signal is broken down into a succession of overlapping finite-duration speech and/or audio signal segments, and a first linear transform is applied to each of these speech and/or audio signal segments to obtain short-term spectral components. Then, a non-linear transform is applied to the short-term spectral components in order to produce warped spectral components. A second linear transform is applied to the warped spectral components to obtain a time-domain signal interval, the time-domain signal interval is multiplied by a time window to produce a windowed-signal interval, and the successive overlapping windowed-signal intervals corresponding to the successive overlapping finite-duration speech and/or audio signal segments are added to obtain a pre-processed signal. This pre-processed signal is encoded to produce the encoded speech and/or audio signal. Corresponding speech/audio decoding method and device are also provided.
摘要翻译：鉴于增强感知质量的方法和装置，用于对语音和/或音频信号进行编码，时域语音和/或音频信号被分解成一系列重叠的有限持续时间语音和/或音频信号段，并且对这些语音和/或音频信号段中的每一个应用第一线性变换以获得短期频谱分量。然后，将非线性变换应用于短期光谱分量，以便产生翘曲的光谱分量。将第二线性变换应用于翘曲频谱分量以获得时域信号间隔，时域信号间隔乘以时间窗以产生窗口信号间隔，并且相应的连续重叠窗口信号间隔对应于连续重叠的有限持续时间语音和/或音频信号段被添加以获得预处理的信号。该预处理信号被编码以产生编码的语音和/或音频信号。还提供了相应的语音/音频解码方法和装置。

8. 发明申请

WO1995028699A1 DIFFERENTIAL-TRANSFORM-CODED EXCITATION FOR SPEECH AND AUDIO CODING 审中-公开
标题翻译：用于语音和音频编码的差分转换编码激活
公开(公告)号：WO1995028699A1
公开(公告)日：1995-10-26
申请号：PCT/CA1995000216
申请日：1995-04-18
申请人： UNIVERSITE DE SHERBROOKE , ADOUL, Jean-Pierre , LAFLAMME, Claude , SALAMI, Redwan , LEFEBVRE, Roch
发明人： UNIVERSITE DE SHERBROOKE
IPC分类号： G10L07/06
CPC分类号： G10L19/038
摘要： A method for coding speech is disclosed. This method, called DTCX (Differential-Transform-Coded Excitation), combines in a new way, the best features of time domain techniques such as CELP (Code-Excited Linear Prediction) with the best features of frequency-domain techniques such as TC (Transform coding) while avoiding their respective drawbacks. The invention preserves the principle of error minimization in the perceptually-weighted-speech(/audio) domain which is found in CELP along with techniques such as linear filtering and pitch prediction, yet, it circumvents the complexity of the CELP analysis-by-synthesis approach by using quantization. The invention also takes advantage of the efficient frequency-domain differential-quantization techniques typical of transform coding (TC) such as spectral decimation, flexible bit allocation as well as numerous forms of stored or algebraic vector quantization techniques. In addition, it is the difference between the current and previous spectra which is quantized resulting in enhanced performance in particular for audio coding. Yet unlike TC, the invention is essentially free from framing problems that plague block transforming of continuous processes.
摘要翻译：公开了一种用于编码语音的方法。称为DTCX（差分变换编码激励）的方法以新的方式组合了时域技术的最佳特征，如CELP（代码激励线性预测），具有频域技术的最佳特征，如TC（变换编码），同时避免其各自的缺点。本发明保留了在CELP中发现的感知加权语音（/音频）域中的误差最小化的原理以及诸如线性滤波和音调预测的技术，但是它避免了CELP综合分析的复杂性通过使用量化的方法。本发明还利用了诸如频谱抽取，灵活位分配以及多种形式的存储或代数矢量量化技术的典型的变换编码（TC）的有效的频域微分量化技术。此外，它是量化的当前和以前的频谱之间的差异，导致特别是用于音频编码的增强的性能。然而与TC不同的是，本发明基本上没有构成对连续过程进行阻塞转换的问题。

9. 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。

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