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。

公开(公告)号：US20160165375A1

公开(公告)日：2016-06-09

申请号：US15041209

申请日：2016-02-11

Applicant: Electronics and Telecommunications Research Institute

Inventor： Jeong-Il SEO ,  Seung-Kwon BEACK ,  Tae-Jin LEE ,  Yong-Ju LEE ,  Dae-Young JANG ,  Kyeongok KANG ,  Jin-Woo HONG ,  Jin-Woong KIM ,  Chieteuk AHN

IPC: H04S7/00 ,  G10L19/008 ,  H04S5/00

CPC classification number: H04S7/308 ,  G10L19/008 ,  H04S5/00 ,  H04S2400/03 ,  H04S2400/11

Abstract: Provided is a method and apparatus for generating a side information bitstream of a multi-object audio signal. The apparatus for generating a side information bitstream of a multi-object audio signal includes a spatial cue information input unit configured to receive spatial cue information generated in an encoder of the multi-object audio signal, a preset information input unit configured to receive preset information for the multi-object audio signal, and a side information bitstream generator configured to generate the side information bitstream based on the spatial cue information and the preset information. The side information bitstream includes a header region and a frame region, and the preset information is included in the frame region.

Abstract translation: 提供了一种用于生成多对象音频信号的侧信息比特流的方法和装置。 用于生成多对象音频信号的侧信息比特流的装置包括：空间提示信息输入单元，被配置为接收在多对象音频信号的编码器中生成的空间提示信息;预设信息输入单元，被配置为接收预设信息 以及侧信息比特流发生器，被配置为基于空间提示信息和预设信息生成侧信息比特流。 侧信息比特流包括报头区域和帧区域，并且预设信息被包括在帧区域中。

公开(公告)号：US20250147715A1

公开(公告)日：2025-05-08

申请号：US18794188

申请日：2024-08-05

Applicant: Electronics and Telecommunications Research Institute

Inventor： Jae-hyoun YOO ,  Kyeongok KANG ,  Yong Ju LEE ,  Dae Young JANG

IPC: G06F3/16

Abstract: A method of playing a sound source is provided. The method includes identifying a playback area for at least one higher order ambisonic (HOA) sound source existing in a target space, determining a coordinate calculation standard for playing the HOA sound source based on a positional relationship between the playback area of the HOA sound source and a listener existing in the target space, and calculating a head related transfer function (HRTF) rendering angle of the HOA sound source according to the determined coordinate calculation standard and playing the HOA sound source.

公开(公告)号：US20250069609A1

公开(公告)日：2025-02-27

申请号：US18948841

申请日：2024-11-15

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

Inventor： Seung Kwon BEACK ,  Tae Jin LEE ,  Min Je KIM ,  Dae Young JANG ,  Kyeongok KANG ,  Jin Woo HONG ,  Ho Chong PARK ,  Young-cheol PARK

IPC: G10L19/02

Abstract: An encoding apparatus and a decoding apparatus in a transform between a Modified Discrete Cosine Transform (MDCT)-based coder and a different coder are provided. The encoding apparatus may encode additional information to restore an input signal encoded according to the MIDCT-based coding scheme, when switching occurs between the MDCT-based coder and the different coder. Accordingly, an unnecessary bitstream may be prevented from being generated, and minimum additional information may be encoded.

公开(公告)号：US20240235511A9

公开(公告)日：2024-07-11

申请号：US18480259

申请日：2023-10-03

Applicant: Electronics and Telecommunications Research Institute

Inventor： Yong Ju LEE ,  Jae-hyoun YOO ,  Dae Young JANG ,  Soo Young PARK ,  Young Ho JEONG ,  Kyeongok KANG ,  Tae Jin LEE

IPC: H03G7/00

CPC classification number: H03G7/007

Abstract: A rendering method of an object-based audio signal and an apparatus for performing the same are provided. The rendering method of an object-based audio signal includes obtaining a rendered audio signal, performing clipping prevention on the rendered audio signal using a first limiter, mixing a signal output by the first limiter using a mixer, and performing clipping prevention on the mixed signal using a second limiter.

公开(公告)号：US20240105194A1

公开(公告)日：2024-03-28

申请号：US18529830

申请日：2023-12-05

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 ,  G10L19/125 ,  G10L19/22 ,  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：US20230362574A1

公开(公告)日：2023-11-09

申请号：US18320729

申请日：2023-05-19

Applicant: Electronics and Telecommunications Research Institute

Inventor： Yong Ju LEE ,  Jae-hyoun YOO ,  Dae Young JANG ,  Kyeongok KANG ,  Soo Young PARK ,  Tae Jin LEE ,  Young Ho JEONG

IPC: H04S7/00 ,  G06F3/16

CPC classification number: H04S7/303 ,  G06F3/165 ,  H04S2420/01 ,  H04S2400/11

Abstract: A method of rendering object-based audio and an electronic device performing the method are disclosed. The method includes identifying metadata of object-based audio, identifying an audio source distance between the object-based audio and a listener using the metadata, determining a minimum distance of the object-based audio to apply attenuation according to the audio source distance, based on a reference distance of the object-based audio in the metadata, and rendering the object-based audio using the audio source distance and the minimum distance.