Electronics and Telecommunications Research Institute

Kwangwoon University Industry-academic Collaboration Foundation

Yonsei University

Wilus Institute of Standards and Technology Inc.

Columbia University

Postech Academy-industry Foundation

University of Southampton

Dimagic Co., Ltd.

Kyeongok Kang

Tae Jin Lee

Dae Young Jang

Jeongil Seo

Jin Woo Hong

Chieteuk Ahn

Young-cheol Park

Min Soo Kim

Jae-hyoun Yoo

Hochong Park

Yong Ju Lee

Jin-Woo Hong

Seungkwon Beack

Seung Kwon Beack

Ho Chong Park

Seung-Kwon Baek

Jinwoong Kim

Min Je Kim

Jin Woong Kim

Hyunoh Oh

Taegyu Lee

Yongju Lee

Daeyoung Jang

Youngcheol Park

Daehee Youn

Tae-Jin Lee

Jin-Woong Kim

Dae-Young Jang

Hwan Shim

Inseon Jang

Hyunjoo Chung

Hyun Sung Chang

Kwangki Kim

Minsoo Hahn

Young Ho Jeong

Seung Kwon Baek

Soo Young Park

Jae-Hyoun Yoo

Young Cheol Park

Koeng-Mo Sung

Hui Yong Kim

Jin Soo Choi

Yong Wang

Jeong-Il Seo

Yong-Ju Lee

Shih-Fu Chang

Jae-Gon Kim

Toshio Saito

Seung-Kwon Beack

Mi Suk Lee

Jooyoung Lee

Philip Arthur Nelson

Minje Kim

Jeho Nam

Eung Don Lee

Hyon-Gon Choo

Hareo Hamada

Seungjun Yang

Seung Jun Yang

Jun-Seok Lim

Young-Cheol Park

Sangwoo Ahn

Hyun-joo Chung

Sang-Kwon Shin

Seungjin Choi

Jiho Yoo

Hyunsung Chang

Filippo Maria Fazi

SeungChul Ham

Jinwoo Hong

Koen-Mo Sung

Seungchul Ham

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.

Rendering method of preventing object-based audio from clipping and apparatus for performing the same

A method of rendering object-based audio and an electronic device for performing the method are disclosed. The method includes identifying metadata of the object-based audio, determining whether the metadata includes a parameter set for an atmospheric absorption effect for each distance, and rendering the object-based audio, using a distance between the object-based audio and a listener obtained using the metadata and the atmospheric absorption effect according to an effect of a medium attenuation based on the parameter, when the metadata includes the parameter.

Method of rendering object-based audio and electronic device for performing the method

An audio signal processing apparatus and an audio signal processing method are disclosed. The audio signal processing method performed by the audio signal processing apparatus includes determining whether a line of sight between a render item (RI) corresponding to an audio element and a listener is visible, based on a bitstream, in response to a case where the line of sight is invisible, generating an audio signal by rendering a diffraction-type RI corresponding to the RI, and outputting the audio signal.

Audio signal processing apparatus and audio signal processing method

A method and apparatus for processing acoustic spatial information are provided. The method of processing acoustic spatial information includes identifying at least one mesh disposed in an acoustic space, setting a minimum cuboid surrounding the mesh as a bounding box, and generating acoustic spatial information including information about the bounding box.

Method and apparatus for processing acoustic spatial information

Provided is a method of processing mesh data for processing an audio signal for audio rendering in a virtual reality (VR) space. The method includes receiving mesh data defining geometry of a certain space and the mesh data includes data regarding three-dimensional (3D) coordinates of points for configuring spatial information of the certain space and processing the mesh data by identifying outermost points among the points.

Method of processing mesh data for processing audio signal for audio rendering in virtual reality space

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.

Unified speech/audio codec (USAC) processing windows sequence based mode switching

Disclosed is an apparatus for immersive spatial audio modeling and rendering for effectively transmitting and playing immersive spatial audio content. The apparatus for immersive spatial audio modeling and rendering disclosed herein may model a spatial audio scene, generate and transmit parameters necessary for spatial audio rendering, and generate various spatial audio effects using the spatial audio parameters, to provide an immersive three-dimensional (3D) audio source coinciding with visual experience in a virtual reality space in response to free changes in the position and direction of a remote user in the space.

Apparatus for immersive spatial audio modeling and rendering

Provided is a method of reproducing an ambisonic signal in a virtual reality (VR) space. The ambisonic signal reproduction method may include receiving an ambisonic signal, mapping the ambisonic signal to channels localized on a sphere according to an equivalent spatial domain (ESD) standard corresponding to an order of the ambisonic signal, and performing a sound field reproduction in the VR space based on the channels localized on the sphere.

Method and apparatus for ambisonic signal reproduction in virtual reality space

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.

Apparatus for encoding and decoding of integrated speech and audio

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 MDCT-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.

Encoding apparatus and decoding apparatus for transforming between modified discrete cosine transform-based coder and different coder

A method and apparatus for rendering an object-based audio signal considering an obstacle are disclosed. A method for rendering an object-based audio signal according to an example embodiment, the method includes identifying an object-based input signal and metadata for the input signal, generating a binaural filter based on the metadata using a binaural room impulse response (BRIR), determining, based on the metadata, whether an obstacle is present between a listener and an object, modifying the generated binaural filter when it is determined that the obstacle is present, and generating a rendered output signal by convolving the modified binaural filter and the input signal.

Method and apparatus for rendering object-based audio signal considering obstacle

The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.

Method and apparatus for binaural rendering audio signal using variable order filtering in frequency domain

A method and system for processing an obstacle effect in a virtual acoustic space are disclosed. The method includes receiving a parameter for an obstacle candidate plane extracted from spatial information, determining, in response to the parameter, whether the obstacle candidate plane is an obstacle related to a path between a position of a virtual sound source and a position of a user, and applying a sound effect according to the obstacle to an audio signal when the obstacle candidate plane is the obstacle. The obstacle candidate plane is a plane of an object that may become the obstacle in a sound propagation path between the virtual sound source and the user.

Method and system for processing obstacle effect in virtual acoustic space

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).

LPC residual signal encoding/decoding apparatus of modified discrete cosine transform (MDCT)-based unified voice/audio encoding device

Growing community of inventors

Daejeon, South Korea

Kyeongok Kang