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2. 发明申请

WO2015157843A1 METHODS, ENCODER AND DECODER FOR LINEAR PREDICTIVE ENCODING AND DECODING OF SOUND SIGNALS UPON TRANSITION BETWEEN FRAMES HAVING DIFFERENT SAMPLING RATES 审中-公开
标题翻译：方法，编码器和解码器用于线性预测编码和解码具有不同采样率的帧之间的转换的声音信号
公开(公告)号：WO2015157843A1
公开(公告)日：2015-10-22
申请号：PCT/CA2014/050706
申请日：2014-07-25
申请人： VOICEAGE CORPORATION
发明人： SALAMI, Redwan , EKSLER, Vaclav
IPC分类号： G10L19/12 , G10L19/06 , G10L19/26
CPC分类号： G10L19/12 , G10L19/06 , G10L19/07 , G10L19/167 , G10L19/173 , G10L19/24 , G10L19/26 , G10L21/038 , G10L25/06 , G10L2019/0002 , G10L2019/0004 , G10L2019/0016
摘要： Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.
摘要翻译：方法，编码器和解码器被配置用于在具有不同内部采样率的帧之间转换。线性预测（LP）滤波器参数从采样率S1转换为采样率S2。使用LP滤波器参数以采样率S1计算LP合成滤波器的功率谱。修改LP合成滤波器的功率谱，将其从采样率S1转换为采样率S2。对LP合成滤波器的修正功率谱进行逆变换，以采样速率S2确定LP合成滤波器的自相关性。自相关用于以采样率S2计算LP滤波器参数。

3. 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。

4. 发明申请

WO2011127569A1 FLEXIBLE AND SCALABLE COMBINED INNOVATION CODEBOOK FOR USE IN CELP CODER AND DECODER 审中-公开
标题翻译：灵活和可扩展的组合创新编码在CELP编码器和解码器中使用的代码
公开(公告)号：WO2011127569A1
公开(公告)日：2011-10-20
申请号：PCT/CA2011/000398
申请日：2011-04-08
申请人： VOICEAGE CORPORATION , BESSETTE, Bruno
发明人： BESSETTE, Bruno
IPC分类号： G10L19/12
CPC分类号： G10L19/12 , G10L19/0212 , G10L19/09 , G10L19/24
摘要： In a CELP coder, a combined innovation codebook coding device comprises a pre-quantizer of a first, adaptive-codebook excitation residual, and a CELP innovation-codebook search module responsive to a second excitation residual produced from the first, adaptive-codebook excitation residual. In a CELP decoder, a combined innovation codebook comprises a de-quantizer of pre-quantized coding parameters into a first excitation contribution, and a CELP innovation-codebook structure responsive to CELP innovation-codebook parameters to produce a second excitation contribution.
摘要翻译：在CELP编码器中，组合创新码本编码装置包括第一自适应码本激励残差的预量化器和响应于从第一自适应码本激励残差产生的第二激励残差的CELP创新码本搜索模块。在CELP解码器中，组合创新码本包括预量化编码参数到第一激励贡献的去量化器，以及响应于CELP创新码本参数以产生第二激励贡献的CELP创新码本结构。

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

WO2010003663A1 AUDIO ENCODER AND DECODER FOR ENCODING FRAMES OF SAMPLED AUDIO SIGNALS 审中-公开
标题翻译：音频编码器和解码器用于编码采样音频信号的框架
公开(公告)号：WO2010003663A1
公开(公告)日：2010-01-14
申请号：PCT/EP2009/004947
申请日：2009-07-08
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , VOICEAGE CORPORATION , LECOMTE, Jérémie , GOURNAY, Philippe , BAYER, Stefan , MULTRUS, Markus , RETTELBACH, Nikolaus
发明人： LECOMTE, Jérémie , GOURNAY, Philippe , BAYER, Stefan , MULTRUS, Markus , RETTELBACH, Nikolaus
IPC分类号： G10L19/14
CPC分类号： G10L19/20
摘要： An audio encoder (100) adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame comprises a number of time domain audio samples, comprising a predictive coding analysis stage (110) for determining information on coefficients of a synthesis filter and information on a prediction domain frame based on a frame of audio samples. The audio encoder (100) further comprises a frequency domain transformer (120) for transforming a frame of audio samples to the frequency domain to obtain a frame spectrum and an encoding domain decider (130). Moreover, the audio encoder (100) comprises a controller (140) for determining an information on a switching coefficient when the encoding domain decider decides that encoded data of a current frame is based on the information on the coefficients and the information on the prediction domain frame when encoded data of a previous frame was encoded based on a previous frame spectrum.
摘要翻译：一种适于编码采样音频信号的帧以获得编码帧的音频编码器（100），其中一帧包括多个时域音频样本，包括用于确定关于合成滤波器的系数的信息的预测编码分析阶段（110）以及基于音频样本帧的关于预测域帧的信息。音频编码器（100）还包括频域变换器（120），用于将音频样本的帧变换到频域以获得帧频和编码域决定器（130）。此外，音频编码器（100）包括控制器（140），用于当编码域决定器基于关于系数的信息和关于预测域的信息来确定当前帧的编码数据时，确定关于切换系数的信息基于前一帧频谱对前一帧的编码数据进行编码的帧。

7. 发明申请

WO2010003491A1 AUDIO ENCODER AND DECODER FOR ENCODING AND DECODING FRAMES OF SAMPLED AUDIO SIGNAL 审中-公开
标题翻译：用于编码和解码采样音频信号的音频编码器和解码器
公开(公告)号：WO2010003491A1
公开(公告)日：2010-01-14
申请号：PCT/EP2009/004015
申请日：2009-06-04
申请人： FRAUNHOFER-GESELLSCHAFT ZUR FÖRDERUNG DER ANGEWANDTEN FORSCHUNG E.V. , VOICEAGE CORPORATION , GEIGER, Ralf , GRILL, Bernhard , BESSETTE, Bruno , GOURNAY, Philippe , FUCHS, Guillaume , MULTRUS, Markus , NEUENDORF, Max , SCHULLER, Gerald
发明人： GEIGER, Ralf , GRILL, Bernhard , BESSETTE, Bruno , GOURNAY, Philippe , FUCHS, Guillaume , MULTRUS, Markus , NEUENDORF, Max , SCHULLER, Gerald
IPC分类号： G06F17/14 , G10L19/02 , G10L19/14
CPC分类号： G10L19/20 , G10L19/0212 , G10L19/08
摘要： An audio encoder (10) adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame comprises a number of time domain audio samples. The audio encoder (10) comprises a predictive coding analysis stage (12) for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder (10) further comprises a time-aliasing introducing transformer (14) for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer (14) is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder (10) comprises a redundancy reducing encoder (16) for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra.
摘要翻译：一种音频编码器（10），适于对采样的音频信号的帧进行编码以获得编码的帧，其中帧包括多个时域音频采样。音频编码器（10）包括用于基于音频样本的帧来确定关于合成滤波器和预测域帧的系数的信息的预测编码分析阶段（12）。音频编码器（10）还包括时间混叠引入变压器（14），用于将重叠的预测域帧变换到频域以获得预测域帧频谱，其中时间混叠引入变压器（14）适于变换重叠预测域帧采用严格抽样方式。此外，音频编码器（10）包括用于根据系数和编码的预测域帧频谱对预测域帧频谱进行编码以获得编码帧的冗余减少编码器（16）。

8. 发明申请

WO2009033288A1 METHOD AND DEVICE FOR FAST ALGEBRAIC CODEBOOK SEARCH IN SPEECH AND AUDIO CODING 审中-公开
标题翻译：用于在语音和音频编码中快速查看代码的方法和设备
公开(公告)号：WO2009033288A1
公开(公告)日：2009-03-19
申请号：PCT/CA2008/001620
申请日：2008-09-11
申请人： VOICEAGE CORPORATION , SALAMI, Redwan , EKSLER, Vaclav , JELINEK, Milan
发明人： SALAMI, Redwan , EKSLER, Vaclav , JELINEK, Milan
IPC分类号： G10L19/00 , G10L19/12 , H04M3/56 , H04N7/15
CPC分类号： G10L19/107
摘要： A method and device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses distributed over the pulse positions. In the algebraic codebook searching method and device, a reference signal for use in searching the algebraic codebook is calculated. In a first stage, a position of a first pulse is determined in relation with the reference signal and among the number of pulse positions. In each of a number of stages subsequent to the first stage, (a) an algebraic codebook gain is recomputed, (b) the reference signal is updated using the recomputed algebraic codebook gain and (c ) a position of another pulse is determined in relation with the updated reference signal and among the number of pulse positions. A codevector of the algebraic codebook is computed using the positions of the pulses determined in the first and subsequent stages, wherein a number of the first and subsequent stages corresponds to the number of pulses in the codevectors of the algebraic codebook.
摘要翻译：一种用于在编码声音信号期间搜索代数码本的方法和装置，其中代数码本包括由多个脉冲位置组成的一组码矢量和分布在脉冲位置上的脉冲数。在代数码本搜索方法和装置中，计算用于搜索代数码本的参考信号。在第一阶段中，相对于参考信号和脉冲位置数确定第一脉冲的位置。在第一阶段之后的多个阶段的每个阶段中，（a）代数码本增益被重新计算，（b）使用重新计算的代数码本增益来更新参考信号，并且（c）另一个脉冲的位置被确定具有更新的参考信号和脉冲位置的数量。使用在第一级和后级中确定的脉冲的位置来计算代数码本的码矢量，其中第一级和后级的数量对应于代数码本的代码矢量中的脉冲数。

9. 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。

10. 发明申请

WO2003102921A1 METHOD AND DEVICE FOR EFFICIENT FRAME ERASURE CONCEALMENT IN LINEAR PREDICTIVE BASED SPEECH CODECS 审中-公开
标题翻译：基于线性预测的语音编码器中有效框架隐藏的方法和装置
公开(公告)号：WO2003102921A1
公开(公告)日：2003-12-11
申请号：PCT/CA2003/000830
申请日：2003-05-30
申请人： VOICEAGE CORPORATION , JELINEK, Milan , GOURNAY, Philippe
发明人： JELINEK, Milan , GOURNAY, Philippe
IPC分类号： G10L19/00
CPC分类号： G10L19/005
摘要： The present invention relates to a method and device for improving concealment of frame erasure caused by frames of an encoded sound signal erased during transmission from an encoder (106) to a decoder (110), and for accelerating recovery of the decoder after non erased frames of the encoded sound signal have been received. For that purpose, concealment/recovery parameters are determined in the encoder or decoder. When determined in the encoder (106), the concealment/recovery parameters are transmitted to the decoder (110). In the decoder, erasure frame concealment and decoder recovery is conducted in response to the concealment/recovery parameters. The concealment/recovery parameters may be selected from the group consisting of: a signal classification parameter, an energy information parameter and a phase information parameter. The determination of the concealment/recovery parameters comprises classifying the successive frames of the encoded sound signal as unvoiced, unvoiced transition, voiced transition, voiced, or onset, and this classification is determined on the basis of at least a part of the following parameters: a normalized correlation parameter, a spectral tilt parameter, a signal-to-noise ratio parameter, a pitch stability parameter, a relative frame energy parameter, and a zero crossing parameter.
摘要翻译：本发明涉及一种用于改善由编码器（106）到解码器（110）的传输期间被擦除的编码声音信号的帧引起的帧擦除隐藏的方法和装置，并且用于在非擦除帧之后加速解码器的恢复已经接收到编码声音信号。为此，在编码器或解码器中确定隐藏/恢复参数。当在编码器（106）中确定时，隐藏/恢复参数被传送到解码器（110）。在解码器中，响应于隐藏/恢复参数进行擦除帧隐藏和解码器恢复。隐藏/恢复参数可以从由信号分类参数，能量信息参数和相位信息参数组成的组中选择。隐藏/恢复参数的确定包括将编码声音信号的连续帧分类为无声，无声转换，有声转换，有声或起始，并且该分类基于以下参数的至少一部分来确定：标准化相关参数，频谱倾斜参数，信噪比参数，音调稳定性参数，相对帧能量参数和过零参数。

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