公开(公告)号：US20190122676A1

公开(公告)日：2019-04-25

申请号：US16224820

申请日：2018-12-19

Applicant: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE ,  INDUSTRY-ACADEMIC COOPERATION FOUNDATION, YONSEI UNIVERSITY ,  WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.

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

IPC: G10L19/008 ,  H04S3/00 ,  H04R5/033

Abstract: 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.

公开(公告)号：US20240212698A1

公开(公告)日：2024-06-27

申请号：US18426726

申请日：2024-01-30

Applicant: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE ,  KWANGWOON UNIVERSITY INDUSTRY-ACADEMIC COLLABORATION FOUNDATION

Inventor： Seungkwon BEACK ,  Tae Jin LEE ,  Min Je KIM ,  Kyeongok KANG ,  Dae Young JANG ,  Jeongil SEO ,  Jin Woo HONG ,  Chieteuk AHN ,  Ho Chong PARK ,  Young-cheol PARK

IPC: G10L19/22 ,  G10L19/022 ,  G10L19/06 ,  G10L19/18

CPC classification number: G10L19/22 ,  G10L19/022 ,  G10L19/06 ,  G10L19/18

Abstract: A Unified Speech and Audio Codec (USAC) that may process a window sequence based on mode switching is provided. The USAC may perform encoding or decoding by overlapping between frames based on a folding point when mode switching occurs. The USAC may process different window sequences for each situation to perform encoding or decoding, and thereby may improve a coding efficiency.

公开(公告)号：US20240119948A1

公开(公告)日：2024-04-11

申请号：US18212364

申请日：2023-06-21

Applicant: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE ,  Kwangwoon University Industry-Academic Collaboration Foundation

Inventor： Tae Jin LEE ,  Seung-Kwon BAEK ,  Min Je KIM ,  Dae Young JANG ,  Jeongil SEO ,  Kyeongok KANG ,  Jin-Woo HONG ,  Hochong PARK ,  Young-Cheol PARK

IPC: G10L19/008 ,  G10L19/02 ,  G10L19/04 ,  G10L19/12 ,  G10L19/20

CPC classification number: G10L19/008 ,  G10L19/02 ,  G10L19/04 ,  G10L19/12 ,  G10L19/20 ,  G10L19/00

Abstract: Provided is an encoding apparatus for integrally encoding and decoding a speech signal and a audio signal, and may include: an input signal analyzer to analyze a characteristic of an input signal; a stereo encoder to down mix the input signal to a mono signal when the input signal is a stereo signal, and to extract stereo sound image information; a frequency band expander to expand a frequency band of the input signal; a sampling rate converter to convert a sampling rate; a speech signal encoder to encode the input signal using a speech encoding module when the input signal is a speech characteristics signal; a audio signal encoder to encode the input signal using a audio encoding module when the input signal is a audio characteristic signal; and a bitstream generator to generate a bitstream.

公开(公告)号：US20220406321A1

公开(公告)日：2022-12-22

申请号：US17895256

申请日：2022-08-25

Applicant: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE ,  KWANGWOON UNIVERSITY INDUSTRY-ACADEMIC COLLABORATION FOUNDATION

Inventor： Seungkwon BEACK ,  Tae Jin LEE ,  Min Je KIM ,  Kyeongok KANG ,  Dae Young JANG ,  Jeongil SEO ,  Jin Woo HONG ,  Chieteuk AHN ,  Ho Chong PARK ,  Young-cheol PARK

IPC: G10L19/22 ,  G10L19/022 ,  G10L19/06 ,  G10L19/18

Abstract: A Unified Speech and Audio Codec (USAC) that may process a window sequence based on mode switching is provided. The USAC may perform encoding or decoding by overlapping between frames based on a folding point when mode switching occurs. The USAC may process different window sequences for each situation to perform encoding or decoding, and thereby may improve a coding efficiency.

公开(公告)号：US20220383581A1

公开(公告)日：2022-12-01

申请号：US17824455

申请日：2022-05-25

Applicant: Electronics and Telecommunications Research Institute

Inventor： Hyun Cheol KIM ,  HYUKMIN KWON ,  Jeongil SEO ,  Sangwoo AHN ,  Seung Jun YANG

IPC: G06T15/04 ,  G06T15/50 ,  G06T17/20

Abstract: Provided are a texture mapping method using a reference image, and a computing apparatus. In detail, a texture mapping method may provide color continuity using a reference image and a target image with respect to each triangular face without analyzing a spatial configuration of a three-dimensional mesh model, thereby reducing a sense of color difference with respect to a texture, and improving quality of the three-dimensional mesh model.

公开(公告)号：US20190074022A1

公开(公告)日：2019-03-07

申请号：US16179120

申请日：2018-11-02

Applicant: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE ,  KWANGWOON UNIVERSITY INDUSTRY-ACADEMIC COLLABORATION FOUNDATION

Inventor： Tae Jin LEE ,  Seung-Kwon BAEK ,  Min Je KIM ,  Dae Young JANG ,  Jeongil SEO ,  Kyeongok KANG ,  Jin-Woo HONG ,  Hochong PARK ,  Young-cheol PARK

IPC: G10L19/12

CPC classification number: G10L19/12 ,  G10L19/008 ,  G10L19/02 ,  G10L19/20 ,  G10L19/22 ,  G11C2207/16

Abstract: Provided are an apparatus and a method for integrally encoding and decoding a speech signal and a audio signal. The encoding apparatus may include: an input signal analyzer to analyze a characteristic of an input signal; a first conversion encoder to convert the input signal to a frequency domain signal, and to encode the input signal when the input signal is a audio characteristic signal; a Linear Predictive Coding (LPC) encoder to perform LPC encoding of the input signal when the input signal is a speech characteristic signal; and a bitstream generator to generate a bitstream using an output

公开(公告)号：US20170337929A1

公开(公告)日：2017-11-23

申请号：US15669262

申请日：2017-08-04

Applicant: Electronics and Telecommunications Research Institute

Inventor： Seung Kwon BEACK ,  Tae Jin LEE ,  Min Je KIM ,  Kyeongok KANG ,  Dae Young JANG ,  Jin Woo HONG ,  Jeongil SEO ,  Chieteuk AHN ,  Hochong PARK ,  Young-cheol PARK

IPC: G10L19/087 ,  G10L19/26

CPC classification number: G10L19/087 ,  G10L19/125 ,  G10L19/22 ,  G10L19/26

Abstract: Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP).

公开(公告)号：US20160314798A1

公开(公告)日：2016-10-27

申请号：US15200404

申请日：2016-07-01

Applicant: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE ,  KWANGWOON UNIVERSITY INDUSTRY-ACADEMIC COLLABORATION FOUNDATION

Inventor： Seungkwon BEACK ,  Tae Jin LEE ,  Min Je KIM ,  Kyeongok KANG ,  Dae Young JANG ,  Jeongil SEO ,  Jin Woo HONG ,  Chieteuk AHN ,  Ho Chong PARK ,  Young-cheol PARK

IPC: G10L19/22 ,  G10L19/022 ,  G10L19/06

CPC classification number: G10L19/22 ,  G10L19/022 ,  G10L19/06 ,  G10L19/18

Abstract: A Unified Speech and Audio Codec (USAC) that may process a window sequence based on mode switching is provided. The USAC may perform encoding or decoding by overlapping between frames based on a folding point when mode switching occurs. The USAC may process different window sequences for each situation to perform encoding or decoding, and thereby may improve a coding efficiency.

公开(公告)号：US20160275956A1

公开(公告)日：2016-09-22

申请号：US15031274

申请日：2014-10-22

Applicant: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE ,  INDUSTRY-ACADEMIC COOPERATION FOUNDATION, YONSEI UNIVERSITY ,  WILUS INSTITUTE OF STANDARDS AND TECHNOLOGY INC.

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

IPC: G10L19/008 ,  H04S3/00

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

Abstract: 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.

Abstract translation: 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为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截止的子带滤波器系数的长度不同 子带 以及通过在相应子带中以预定的块大小快速傅里叶变换（FFT）所述截断的子带滤波器系数并使用该FFT滤波器系数的参数化单元来生成FFT滤波器系数。