公开(公告)号：WO2016173658A1

公开(公告)日：2016-11-03

申请号：PCT/EP2015/059476

申请日：2015-04-30

Applicant: HUAWEI TECHNOLOGIES CO., LTD. ,  SETIAWAN, Panji ,  HELWANI, Karim

Inventor： SETIAWAN, Panji ,  HELWANI, Karim

IPC: G10L19/008

CPC classification number: G10L19/008 ,  H04S3/008 ,  H04S2400/01 ,  H04S2400/03

Abstract: The invention relates to audio signal processing apparatuses and methods, such as an audio signal downmixing apparatus (105) for processing an input audio signal comprising a plurality of input channels (113) into an output audio signal comprising a plurality of primary output channels (123) and at least one auxiliary output channel (125) using a downmix matrix D, wherein the downmix matrix D comprises a primary downmix matrix D u providing the plurality of primary output channels (123) and an auxiliary downmix matrix D w providing the at least one auxiliary output channel (125). The audio signal downmixing apparatus (105) comprises an auxiliary downmix matrix determiner (107) configured to determine the auxiliary downmix matrix D w by computing a plurality of eigenvectors of a covariance matrix COV defined by the plurality of input channels (113) of the input audio signal, determining for at least one eigenvector of the plurality of eigenvectors of the covariance matrix COV a subspace angle between the at least one eigenvector and a vector defined by a column of the primary downmix matrix D u , selecting at least one eigenvector from the plurality of eigenvectors based on the subspace angle and a preset threshold angle Θ ΜΙΝ , and defining at least one column of the auxiliary downmix matrix D w by the at least one selected eigenvector, and a processor (109) configured to process the input audio signal into the output audio signal using the downmix matrix D.

Abstract translation: 本发明涉及音频信号处理设备和方法，例如用于将包括多个输入通道（113）的输入音频信号处理成包括多个主输出通道（123）的输出音频信号的音频信号降混装置（105） ）和使用下混合矩阵D的至少一个辅助输出通道（125），其中下混合矩阵D包括提供多个主输出通道（123）的主下混合矩阵Du和提供至少一个辅助的辅助下混合矩阵Dw 输出通道（125）。 音频信号下混合装置（105）包括辅助下混合矩阵确定器（107），其被配置为通过计算由输入音频的多个输入声道（113）定义的协方差矩阵COV的多个特征向量来确定辅助下混矩阵Dw 确定所述协方差矩阵COV的所述多个特征向量中的至少一个特征向量，所述至少一个特征向量与由所述主下混矩阵Du的列定义的向量之间的子空间角，从所述多个子特征向量中选择至少一个特征向量 基于所述子空间角和预设阈值角ΘM N N N的特征向量，以及通过所述至少一个所选择的本征向量定义所述辅助下混合矩阵Dw的至少一列;以及处理器（109），被配置为将所述输入音频信号处理为输出音频 信号使用下混矩阵D.

公开(公告)号：WO2016146180A1

公开(公告)日：2016-09-22

申请号：PCT/EP2015/055637

申请日：2015-03-18

Applicant: HUAWEI TECHNOLOGIES CO., LTD. ,  HELWANI, Karim ,  KONDRAD, Lukasz ,  PIOTTO, Nicola

Inventor： HELWANI, Karim ,  KONDRAD, Lukasz ,  PIOTTO, Nicola

IPC: H04N17/00 ,  H04N9/73 ,  H04N1/60 ,  H04N5/232 ,  H04N13/00

CPC classification number: G06T7/97 ,  G06T7/90 ,  G06T2207/10024 ,  G06T2207/10152 ,  G06T2207/20221 ,  H04N1/486 ,  H04N1/6077 ,  H04N1/6086 ,  H04N5/247 ,  H04N9/735 ,  H04N13/243 ,  H04N13/246 ,  H04N13/257 ,  H04N17/002

Abstract: An image processing device (10) performing color balancing of a first image (11a) and at least a second image (12a) is provided. The image processing device (10) comprises a color balancing determination unit (20) and a color balancing calculation unit (21). The color balancing determination unit (20) determines a global gain vector (t) comprising at least two gain factors (â n , â n+1 ) of the first and second images (11a, 12a) by minimizing a cost function based upon reference pixel values of the first and second images (11a, 12a). The first and second image reference pixels depict a shared color scene of the two images. The color balancing calculation unit (21) performs color balancing of the first image (11a) based upon the gain factor (an) of the first image (11a) and to perform a color balancing of the second image (12a) based upon the gain factor (â n+1 ) of the second image (12a).

Abstract translation: 提供执行第一图像（11a）和至少第二图像（12a）的颜色平衡的图像处理装置（10）。 图像处理装置（10）包括颜色平衡确定单元（20）和颜色平衡计算单元（21）。 颜色平衡确定单元（20）通过使基于参考像素值的成本函数最小化来确定包括第一和第二图像（11a，12a）的至少两个增益因子（Λ+ 1）的全局增益矢量（t） 的第一和第二图像（11a，12a）。 第一和第二图像参考像素描绘两个图像的共享颜色场景。 颜色平衡计算单元（21）基于第一图像（11a）的增益因子（a）执行第一图像（11a）的颜色平衡，并且基于增益来执行第二图像（12a）的颜色平衡 因子（â+ 1）的第二图像（12a）。

公开(公告)号：WO2015058818A1

公开(公告)日：2015-04-30

申请号：PCT/EP2013/073931

申请日：2013-11-15

Applicant: HUAWEI TECHNOLOGIES CO., LTD. ,  FONTANA, Simone ,  HELWANI, Karim ,  GROSCHE, Peter

Inventor： FONTANA, Simone ,  HELWANI, Karim ,  GROSCHE, Peter

IPC: H04S3/00

CPC classification number: H04S7/306 ,  G10L19/008 ,  H04S3/004 ,  H04S2400/01 ,  H04S2400/03 ,  H04S2420/01 ,  H04S2420/07

Abstract: An apparatus and a method for compressing a set of N binaural room impulse responses, BRIR, wherein each channel of an N channel audio signal (I1, I2,..., IN) is convolved with the corresponding compressed set of N BRIR.

Abstract translation: 一种用于压缩一组N个双耳室脉冲响应BRIR的装置和方法，其中N个声道音频信号（I1，I2，...，IN）的每个通道与相应的N BRIR的压缩集进行卷积。

公开(公告)号：WO2017084704A1

公开(公告)日：2017-05-26

申请号：PCT/EP2015/076954

申请日：2015-11-18

Applicant: HUAWEI TECHNOLOGIES CO., LTD. ,  GROSCHE, Peter ,  HELWANI, Karim

Inventor： GROSCHE, Peter ,  HELWANI, Karim ,  SCHÖRKHUBER, Christian ,  ZOTTER, Franz ,  HÖLDRICH, Robert ,  FRANK, Matthias

IPC: H03G3/20 ,  H04R3/00

CPC classification number: H04R3/005 ,  H03G3/3005 ,  H04R2430/01

Abstract: The invention relates to a sound signal processing apparatus (100) for enhancing a sound signal from a target source. The sound signal processing apparatus (100) comprises a plurality of microphones (101a-f), wherein each microphone (101a-f) is configured to receive the sound signal from the target source; an estimator (103) configured to estimate a first power measure on the basis of the sound signal from the target source received by a first microphone (101a-f) of the plurality of microphones (101a-f) and a second power measure on the basis of the sound signal from the target source received by at least a second microphone (101a-f) of the plurality of microphones (101a-f), which is located more distant from the target source than the first microphone (101a-f), wherein the estimator (103) is further configured to determine a gain factor on the basis of a ratio between the second power measure and the first power measure; and an amplifier (105) configured to apply the gain factor to the sound signal from the target source received by the first microphone (101a-f).

Abstract translation: 本发明涉及用于增强来自目标源的声音信号的声音信号处理设备（100）。 声音信号处理设备（100）包括多个麦克风（101a-f），其中每个麦克风（101a-f）被配置为从目标源接收声音信号; 估计器（103），其被配置为基于由所述多个麦克风（101a-f）中的第一麦克风（101a-f）接收的来自所述目标源的声音信号以及所述第二麦克风 基于由多个麦克风（101a-f）中的至少第二麦克风（101a-f）接收的来自目标源的声音信号，该第二麦克风与第一麦克风（101a-f）相比距离目标源更远， 其中，所述估计器（103）还被配置为基于所述第二功率测量与所述第一功率测量之间的比率来确定增益因子; 以及放大器（105），被配置为将所述增益因子应用于由所述第一麦克风（101a-f）接收的来自所述目标源的所述声音信号。

公开(公告)号：WO2015165539A1

公开(公告)日：2015-11-05

申请号：PCT/EP2014/058913

申请日：2014-04-30

Applicant: HUAWEI TECHNOLOGIES CO., LTD. ,  HELWANI, Karim ,  PANG, Liyun

Inventor： HELWANI, Karim ,  PANG, Liyun

IPC: G10L21/0208

CPC classification number: G10L21/0232 ,  G10L19/008 ,  G10L21/0208 ,  G10L2021/02082

Abstract: The invention relates to a signal processing apparatus (100) for dereverberating a number of input audio signals, the signal processing apparatus (100) comprising a transformer (101) being configured to transform the number of input audio signals into a transformed domain to obtain input transformed coefficients, the input transformed coefficients being arranged to form an input transformed coefficient matrix, a filter coefficient determiner (103) being configured to determine filter coefficients upon the basis of eigenvalues resulting from the decomposition of an input auto-coherence matrix, the filter coefficients being arranged to form a filter coefficient matrix, a filter (105) being configured to convolve input transformed coefficients of the input transformed coefficient matrix by filter coefficients of the filter coefficient matrix to obtain output transformed coefficients, the output transformed coefficients being arranged to form an output transformed coefficient matrix, and an inverse transformer (107) being configured to inversely transform the output transformed coefficient matrix from the transformed domain to obtain a number of output audio signals.

Abstract translation: 本发明涉及一种用于对多个输入音频信号进行去混响的信号处理设备（100），该信号处理设备（100）包括变压器（101），其被配置为将输入音频信号的数量变换成变换域以获得输入 输入变换系数被布置以形成输入变换系数矩阵，滤波器系数确定器（103）被配置为基于由输入自相关矩阵的分解得到的特征值来确定滤波器系数，滤波器系数 布置成形成滤波器系数矩阵，滤波器（105）被配置为通过滤波器系数矩阵的滤波器系数来卷积输入变换系数矩阵的输入变换系数，以获得输出变换系数，输出变换系数被布置成形成 输出变换系数矩阵，a n逆变换器（107）被配置为从变换的域逆变换输出变换系数矩阵以获得输出音频信号的数量。

公开(公告)号：WO2017108097A1

公开(公告)日：2017-06-29

申请号：PCT/EP2015/080972

申请日：2015-12-22

Applicant: HUAWEI TECHNOLOGIES DUESSELDORF GMBH ,  HELWANI, Karim ,  CHEN, Kainan ,  GEIGER, Jürgen

Inventor： HELWANI, Karim ,  CHEN, Kainan ,  GEIGER, Jürgen

IPC: G10L21/028 ,  G10L21/0272 ,  G01S3/80 ,  G01S3/808 ,  H04S7/00

CPC classification number: G10L21/028 ,  G01S3/8006 ,  G01S3/8083 ,  H04R1/406 ,  H04R3/005

Abstract: The proposed method for localizing a target sound source from a plurality of sound sources, wherein a multi-channel recording signal of the plurality of sound sources comprises a plurality of microphone channel signals, comprises converting each microphone channel signal into a respective channel spectrogram in a time-frequency domain, blindly separating the channel spectrograms to obtain a plurality of separated source signals, identifying, among the plurality of separated source signals, the separated source signal that best matches a target source model, estimating, based on the identified separated source signal, a binary mask reflecting where the target sound source is active in the channel spectrograms in terms of time and frequency, applying the binary mask on the channel spectrograms to obtain masked channel spectrograms, and localizing the target sound source from the plurality of sound sources based on the masked channel spectrograms.

Abstract translation: 所提出的用于定位来自多个声源的目标声源的方法，其中多个声源的多声道记录信号包括多个麦克风声道信号，包括将每个麦克风声道 信号转换成时频域中的相应信道频谱图，盲分离信道频谱图以获得多个分离的源信号，识别多个分离的源信号中与目标源模型最佳匹配的分离源信号，估计 基于所识别的分离的源信号确定反映目标声源在时间和频率方面在信道谱图中是活动的二进制掩码，在信道谱图上应用二进制掩码以获得掩蔽的信道谱图，以及定位目标声音 来自多个声源的基于掩蔽的声道频谱图的信号源。

公开(公告)号：WO2016173659A1

公开(公告)日：2016-11-03

申请号：PCT/EP2015/059477

申请日：2015-04-30

Applicant: HUAWEI TECHNOLOGIES CO., LTD. ,  SETIAWAN, Panji ,  HELWANI, Karim

Inventor： SETIAWAN, Panji ,  HELWANI, Karim

IPC: G10L19/008

CPC classification number: G10L19/008 ,  H04S3/008 ,  H04S2400/01 ,  H04S2400/03

Abstract: The invention relates to audio signal processing apparatuses and methods, such as an audio signal downmixing apparatus (105) for processing an input audio signal into an output audio signal, wherein the input audio signal comprises a plurality of input channels (113) recorded at a plurality of spatial positions and the output audio signal comprises a plurality of primary output channels (123). The audio signal downmixing apparatus (105) comprises a downmix matrix determiner (107) configured to determine for each frequency bin j of a plurality of frequency bins a downmix matrix DU with j being an integer in the range from 1 to N, wherein for a given frequency bin j the downmix matrix DU maps a plurality of Fourier coefficients associated with the plurality of input channels (113) of the input audio signal into a plurality of Fourier coefficients of the primary output channels (123) of the output audio signal, wherein for frequency bins with j being smaller than or equal to a cutoff frequency bin k the downmix matrix DU is determined by determining eigenvectors of the discrete Laplace-Beltrami operator L defined by the plurality of spatial positions where the plurality of input channels (113) are recorded, and wherein for frequency bins with j being larger than the cutoff frequency bin k the downmix matrix DU is determined by determining a first subset of eigenvectors of a covariance matrix COV defined by the plurality of input channels (113) of the input audio signal, and a processor (109) configured to process the input audio signal using the downmix matrix DU into the output audio signal.

Abstract translation: 本发明涉及音频信号处理装置和方法，例如用于将输入音频信号处理成输出音频信号的音频信号降混装置（105），其中输入音频信号包括多个输入声道（113） 多个空间位置和输出音频信号包括多个主要输出通道（123）。 音频信号降混装置（105）包括下混合矩阵确定器（107），其被配置为为多个频率仓的每个频率仓j确定下混合矩阵DU，其中j为1至N的整数，其中对于 给定频率bin j，下混合矩阵DU将与输入音频信号的多个输入通道（113）相关联的多个傅里叶系数映射成输出音频信号的主要输出通道（123）的多个傅立叶系数，其中 对于j小于或等于截止频率bin k的频率仓，下混矩阵DU通过确定由多个输入通道（113）为多个的多个空间位置定义的离散拉普拉斯 - 贝拉米利算子L的特征向量来确定 并且其中对于具有大于截止频率bin k的j的频率仓，下混合矩阵DU通过确定cov的特征向量的第一子集来确定 由输入音频信号的多个输入声道（113）定义的声音矩阵COV以及配置成使用下混矩阵DU将输入音频信号处理成输出音频信号的处理器（109）。

公开(公告)号：WO2016074734A1

公开(公告)日：2016-05-19

申请号：PCT/EP2014/074536

申请日：2014-11-13

Applicant: HUAWEI TECHNOLOGIES CO., LTD. ,  HELWANI, Karim ,  GROSCHE, Peter ,  LANG, Yue

Inventor： HELWANI, Karim ,  GROSCHE, Peter ,  LANG, Yue

IPC: H04S7/00

CPC classification number: H04S7/303 ,  G06F3/012 ,  H04R5/02 ,  H04R5/04 ,  H04R2499/11 ,  H04S3/00 ,  H04S3/008 ,  H04S5/005 ,  H04S7/00 ,  H04S7/30 ,  H04S7/305 ,  H04S2400/01 ,  H04S2420/01

Abstract: An audio signal processing device (10) for generating a plurality of output signals for a plurality of loudspeakers from an input audio signal comprises a driving function determining unit (11) adapted to determined driving functions of a plurality of loudspeakers for generating a virtual left binaural signal source and a virtual right binaural signal source based upon a position and a directivity of the virtual left binaural signal source, a position and a directivity of the virtual right binaural signal source and positions of the plurality of loudspeakers. Moreover, it comprises a filtering unit (12) adapted to filter a left binaural signal and a right binaural signal using the driving functions of the plurality of loudspeakers resulting in the plurality of output signals. The left binaural signal and the right binaural signal constitute the input audio signal or are derived there from.

Abstract translation: 一种用于从输入音频信号产生多个扬声器的多个输出信号的音频信号处理装置（10）包括适于确定多个扬声器的驱动功能的驱动功能确定单元（11），用于产生虚拟左双耳 信号源和虚拟右双耳信号源，基于虚拟左双耳信号源的位置和方向性，虚拟右双耳信号源的位置和方向性以及多个扬声器的位置。 此外，它包括适于使用多个扬声器的驱动功能来过滤左双耳信号和右双耳信号的滤波单元（12），导致多个输出信号。 左双耳信号和右双耳信号构成输入音频信号或从中导出。