公开(公告)号：EP3062535B1

公开(公告)日：2019-07-03

申请号：EP14856742.3

申请日：2014-10-22

申请人： Industry-Academic Cooperation Foundation, Yonsei University ,  Wilus Institute of Standards and Technology Inc. ,  Electronics and Telecommunications Research Institute

发明人： LEE, Taegyu ,  OH, Hyunoh ,  PARK, Youngcheol ,  YOUN, Daehee ,  SEO, Jeongil ,  LEE, Yongju ,  BEACK, Seungkwon ,  KANG, Kyeongok ,  JANG, Daeyoung

IPC分类号： H04S3/00

公开(公告)号：EP3062534A1

公开(公告)日：2016-08-31

申请号：EP14855415.7

申请日：2014-10-22

申请人： Electronics and Telecommunications Research Institute ,  Industry-Academic Cooperation Foundation, Yonsei University ,  Wilus Institute of Standards and Technology Inc.

发明人： LEE, Taegyu ,  OH, Hyunoh ,  SEO, Jeongil ,  LEE, Yongju ,  BEACK, Seungkwon ,  KANG, Kyeongok ,  JANG, Daeyoung ,  PARK, Youngcheol ,  YOUN, Daehee

IPC分类号： H04S3/00

CPC分类号： G10L19/008 ,  H04R3/00 ,  H04R5/033 ,  H04S3/00 ,  H04S3/002 ,  H04S3/004 ,  H04S3/008 ,  H04S2400/01 ,  H04S2420/03

摘要： The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method for generating a filter for an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity and a parameterization apparatus therefor.
To this end, provided are a method for generating a filter of an audio signal, including: receiving at least one proto-type filter coefficient for filtering each subband signal of an input audio signal; converting the proto-type filter coefficient into a plurality of subband filter coefficients; truncating each of the subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and generating FFT filter coefficients by fast Fourier transforming (FFT) the truncated subband filter coefficients by a predetermined block size in the corresponding subband and a parameterization unit using the same.

摘要翻译： 本发明涉及用于处理信号的方法和装置，其用于有效地再现音频信号，更具体地说，涉及一种用于产生音频信号的滤波器的方法，该方法和装置用于实现用于输入的滤波 具有低计算复杂度的音频信号及其参数化装置。 为此，提供了一种用于产生音频信号的滤波器的方法，包括：接收用于滤波输入音频信号的每个子带信号的至少一个原型滤波器系数; 将原型滤波器系数转换成多个子带滤波器系数; 基于通过至少部分地使用从相应的子带滤波器系数提取的特征信息而获得的滤波器顺序信息来截断每个子带滤波器系数，至少一个截断的子带滤波器系数的长度与另一个子带滤波器系数的截短的子带滤波器系数的长度不同 子带; 以及通过快速傅里叶变换（FFT）在对应子带中的预定块大小的截断子带滤波器系数和使用该滤波器系数的参数化单元来生成FFT滤波器系数。

公开(公告)号：EP3062535A1

公开(公告)日：2016-08-31

申请号：EP14856742.3

申请日：2014-10-22

申请人： Industry-Academic Cooperation Foundation, Yonsei University ,  Wilus Institute of Standards and Technology Inc. ,  Electronics and Telecommunications Research Institute

发明人： LEE, Taegyu ,  OH, Hyunoh ,  PARK, Youngcheol ,  YOUN, Daehee ,  SEO, Jeongil ,  LEE, Yongju ,  BEACK, Seungkwon ,  KANG, Kyeongok ,  JANG, Daeyoung

IPC分类号： H04S3/00

CPC分类号： G10L19/008 ,  G10H2250/111 ,  G10H2250/145 ,  H04R3/00 ,  H04R5/033 ,  H04S3/00 ,  H04S3/002 ,  H04S3/004 ,  H04S3/008 ,  H04S2400/01 ,  H04S2420/03

摘要： The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity.
To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving truncated subband filter coefficients for filtering each subband signal of the input audio signal, the truncated subband filter coefficients being at least a portion of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal, the lengths of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, and the truncated subband filter coefficients being constituted by at least one FFT filter coefficient in which fast Fourier transform (FFT) by a predetermined block size in the corresponding subband has been performed; performing the fast Fourier transform of the subband signal based on a predetermined subframe size in the corresponding subband; generating a filtered subframe by multiplying the fast Fourier transformed subframe and the FFT filter coefficients; inverse fast Fourier transforming the filtered subframe; and generating a filtered subband signal by overlap-adding at least one subframe which is inverse fast Fourier transformed and an apparatus for processing an audio signal using the same.

摘要翻译： 用于处理信号的方法和设备技术领域本发明涉及一种用于处理信号的方法和设备，其用于有效地再现音频信号，并且更具体地涉及用于处理音频信号的方法和设备，其用于实现输入的滤波 具有低计算复杂度的音频信号。 为此，提供了一种用于处理音频信号的方法，包括：接收输入音频信号; 接收用于对输入音频信号的每个子带信号进行滤波的截短的子带滤波器系数，截断的子带滤波器系数是从用于输入音频信号的双耳滤波的双耳室内脉冲响应（BRIR）滤波器系数获得的子带滤波器系数的至少一部分， 截断的子带滤波器系数的长度是基于通过至少部分地使用从相应的子带滤波器系数提取的特征信息而获得的滤波器顺序信息来确定的，并且截断的子带滤波器系数由至少一个FFT滤波器系数构成，其中快速傅立叶 已经执行了相应子频带中的预定块大小的变换（FFT）; 基于相应子频带中的预定子帧大小来执行子带信号的快速傅立叶变换; 通过将快速傅立叶变换的子帧与FFT滤波器系数相乘来产生经滤波的子帧; 对经滤波的子帧进行快速傅立叶逆变换; 以及通过叠加至少一个快速傅立叶逆变换后的子帧来生成滤波后的子带信号，以及使用该子帧信号处理音频信号的装置。

公开(公告)号：EP4246513A3

公开(公告)日：2023-12-13

申请号：EP23190101.8

申请日：2014-12-23

申请人： Wilus Institute of Standards and Technology Inc. ,  GCoA Co., Ltd.

发明人： OH, Hyunoh ,  LEE, Taegyu

IPC分类号： H04S3/00 ,  H04S7/00 ,  H03H17/02 ,  G10L19/008 ,  G10L19/02

摘要： The present invention relates to an audio signal processing method, a parameterization device and an audio signal processing device for the same, and more particularly, to an audio signal processing method to implement filtering of an input audio signal with a low computational complexity, and a parameterization device and an audio signal processing device for the same.
To this end, provided are a method for processing an audio signal, including: receiving an input audio signal; receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the truncated subband filter coefficients being filter coefficients of which energy compensation is performed based on the flag information and the length of at least one truncated subband filter coefficients being different from the length of the truncated subband filter coefficients of another subband; and filtering each subband signal of the input audio signal by using the truncated subband filter coefficients, and a parameterization device and an audio signal processing device for the same.

公开(公告)号：EP3934283A1

公开(公告)日：2022-01-05

申请号：EP21191439.5

申请日：2014-12-23

申请人： Wilus Institute of Standards and Technology Inc. ,  GCoA Co., Ltd.

发明人： OH, Hyunoh ,  LEE, Taegyu

IPC分类号： H04S3/00 ,  H04S7/00 ,  H03H17/02 ,  G10L19/008 ,  G10L19/02

摘要： The present invention relates to an audio signal processing method, a parameterization device and an audio signal processing device for the same, and more particularly, to an audio signal processing method to implement filtering of an input audio signal with a low computational complexity, and a parameterization device and an audio signal processing device for the same.
To this end, provided are a method for processing an audio signal, including: receiving an input audio signal; receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the truncated subband filter coefficients being filter coefficients of which energy compensation is performed based on the flag information and the length of at least one truncated subband filter coefficients being different from the length of the truncated subband filter coefficients of another subband; and filtering each subband signal of the input audio signal by using the truncated subband filter coefficients, and a parameterization device and an audio signal processing device for the same.

公开(公告)号：EP3767970A1

公开(公告)日：2021-01-20

申请号：EP20195668.7

申请日：2014-09-17

申请人： Wilus Institute of Standards and Technology Inc.

发明人： OH, Hyunoh ,  LEE, Taegyu

IPC分类号： H04S3/00 ,  H04S3/02 ,  G10L19/008 ,  G10L19/02 ,  H03H17/02 ,  G10L25/48

摘要： The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount.
To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.

公开(公告)号：EP3048815A1

公开(公告)日：2016-07-27

申请号：EP14846160.1

申请日：2014-09-17

申请人： Wilus Institute of Standards and Technology Inc.

发明人： OH, Hyunoh ,  LEE, Taegyu

IPC分类号： H04S3/00

摘要： The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.
To this end, provided are a method for processing an audio signal including: receiving multi-audio signals including multi-channel or multi-object signals; receiving truncated subband filter coefficients for filtering the multi-audio signals, the truncated subband filter coefficients being at least a portion of subband filter coefficients obtained from a binaural room impulse response (BRIR) filter coefficient for binaural filtering of the multi-audio signals, and the lengths of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients; and filtering the subband signal by using the truncated subband filter coefficients corresponding to each subband signal of the multi-audio signals and an apparatus for processing an audio signal using the same.

摘要翻译： 用于处理信号的方法和设备技术领域本发明涉及用于有效地再现音频信号的信号处理方法和设备，并且更具体地涉及用于处理信号的方法和设备， 立体声通道或多对象音频信号，计算量较小。 为此，提供了一种用于处理音频信号的方法，包括：接收包括多声道或多声道信号的多音频信号; 接收用于对所述多音频信号进行滤波的截短的子带滤波器系数，所述截断的子带滤波器系数是从用于所述多音频信号的双耳滤波的双耳室内脉冲响应（BRIR）滤波器系数获得的子带滤波器系数的至少一部分，以及 截断的子带滤波器系数的长度基于通过至少部分地使用从相应的子带滤波器系数提取的特征信息获得的滤波器顺序信息来确定; 以及使用与多音频信号的每个子带信号对应的截断子带滤波器系数对子带信号进行滤波以及使用该装置对音频信号进行处理。

公开(公告)号：EP3048815B1

公开(公告)日：2020-11-11

申请号：EP14846160.1

申请日：2014-09-17

申请人： Wilus Institute of Standards and Technology Inc.

发明人： OH, Hyunoh ,  LEE, Taegyu

IPC分类号： H04S3/00 ,  G10L19/02 ,  G10L19/008 ,  G10L25/48 ,  H03H17/02

公开(公告)号：EP3697109A1

公开(公告)日：2020-08-19

申请号：EP20168164.0

申请日：2014-12-23

申请人： Wilus Institute of Standards and Technology Inc.

发明人： OH, Hyunoh ,  LEE, Taegyu

IPC分类号： H04S3/00 ,  H04S7/00 ,  H03H17/02 ,  G10L19/008 ,  G10L19/02

摘要： The present invention relates to an audio signal processing method, a parameterization device and an audio signal processing device for the same, and more particularly, to an audio signal processing method to implement filtering of an input audio signal with a low computational complexity, and a parameterization device and an audio signal processing device for the same.
To this end, provided are a method for processing an audio signal, including: receiving an input audio signal; receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the truncated subband filter coefficients being filter coefficients of which energy compensation is performed based on the flag information and the length of at least one truncated subband filter coefficients being different from the length of the truncated subband filter coefficients of another subband; and filtering each subband signal of the input audio signal by using the truncated subband filter coefficients, and a parameterization device and an audio signal processing device for the same.

公开(公告)号：EP3089483A1

公开(公告)日：2016-11-02

申请号：EP14875534.1

申请日：2014-12-23

申请人： Wilus Institute of Standards and Technology Inc.

发明人： OH, Hyunoh ,  LEE, Taegyu

IPC分类号： H04S5/00 ,  H04S7/00

摘要： The present invention relates to an audio signal processing method, a parameterization device and an audio signal processing device for the same, and more particularly, to an audio signal processing method to implement filtering of an input audio signal with a low computational complexity, and a parameterization device and an audio signal processing device for the same.
To this end, provided are a method for processing an audio signal, including: receiving an input audio signal; receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the truncated subband filter coefficients being filter coefficients of which energy compensation is performed based on the flag information and the length of at least one truncated subband filter coefficients being different from the length of the truncated subband filter coefficients of another subband; and filtering each subband signal of the input audio signal by using the truncated subband filter coefficients, and a parameterization device and an audio signal processing device for the same.

摘要翻译： 音频信号处理方法，参数化装置和音频信号处理装置技术领域本发明涉及一种音频信号处理方法，参数化装置和音频信号处理装置，更具体地说，涉及一种以低计算复杂度对输入音频信号进行滤波的音频信号处理方法， 参数化设备和音频信号处理设备。 为此，提供了一种用于处理音频信号的方法，包括：接收输入音频信号; 接收至少一个双耳室脉冲响应（BRIR）滤波器系数用于输入音频信号的双耳滤波; 将BRIR滤波器系数转换成多个子带滤波器系数; 获取指示时域中的BRIR滤波器系数的长度是否大于预定值的标志信息; 基于通过至少部分地使用从相应的子带滤波器系数提取的特征信息获得的滤波器顺序信息来截断每个子带滤波器系数，截断的子带滤波器系数是基于标志信息和at的长度执行能量补偿的滤波器系数 至少一个截断子带滤波器系数与另一子带的截短子带滤波器系数的长度不同; 并且使用截断的子带滤波器系数对输入音频信号的每个子带信号进行滤波，以及用于其的参数化装置和音频信号处理装置。