Wilus Institute of Standards and Technology Inc.

Gaudio Lab, Inc.

Electronics and Telecommunications Research Institute

Yonsei University

Intellectual Discovery Co., Ltd.

Gcoa Co., Ltd.

Naver Corporation

Gaudi Audio Lab, Inc.

Taegyu Lee

Hyunoh Oh

Jeonghun Seo

Hyun Oh Oh

Sangbae Chon

Yonghyun Baek

Jinsam Kwak

Juhyung Son

Jeongil Seo

Kyeongok Kang

Seungkwon Beack

Sewoon Jeon

Yongju Lee

Jeongook Song

Youngcheol Park

Daehee Youn

Daeyoung Jang

Myungsuk Song

Kyutae Park

Jaesung Choi

Younghoon Kwon

Hyunjoo Chung

Byung Gyou Choi

Jiwon Oh

Seongcheol Jo

JiUng Lee

Hyeonsu Seo

Jeongsoo Hur

Taek Su An

Sung Hyun Kang

Byoungjoon Ahn

Dong Hwan Kim

Dae Hwang Kim

Jung Sik Kim

Hyung Oh Oh

Cheolwoo Jeong

A computer system for rendering event-customized audio content and a method thereof. The computer system may be configured to receive an event list of a plurality of events, an audio list for audio rendering information that matches each of the events, and event state information on a current event state, and to output current audio content according to audio rendering information that matches one of the events based on the current event state.

Computer system for rendering event-customized audio content, and method thereof

A method for normalizing platform-adaptive audio includes encoding input video content and generating video stream data as original data to store the video stream data in storage; generating loudness metadata for audio data of the video content and storing the loudness metadata in the storage; receiving a request for the video content from a client; searching the storage for video stream data of the video content corresponding to the request, the loudness metadata, and a device profile corresponding to device information included in the request; and transmitting, to the client, a response including the video stream data, the loudness metadata, and the device profile that are found in the storage.

Method and system for normalizing platform-adaptive audio

An audio signal processing device comprises: a receiver for receiving an input audio signal; a processor for generating loudness metadata corresponding to the input audio signal; and an outputter for transmitting the loudness metadata generated by the processor. The processor is configured to acquire loudness information analyzed from input content, acquires loudness information about the input audio signal by measuring the loudness of the input audio signal, generates the loudness metadata by converting the loudness information, and transmits, through the outputter, the generated loudness metadata to an output device for outputting the input audio signal.

Audio signal processing method and device for controlling loudness level

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 loudness normalization method includes receiving data for playback of content from a server in response to a user's request to play back the content; normalizing the loudness of the content by adjusting the volume output level of a player using volume level metadata of the content included in the received data; and providing the content by playing audio of the content based on the adjusted volume output level of the player.

Loudness normalization method and system

Disclosed is an audio signal processing method including: receiving a stereo signal; transforming the stereo signal into a frequency-domain signal; rendering the first signal based on a first ipsilateral filter coefficient; generating a frontal ipsilateral signal relating to the frequency-domain signal; rendering the second signal based on a second ipsilateral filter coefficient; generating a side ipsilateral signal relating to the frequency-domain signal; rendering the second signal based on a contralateral filter coefficient; generating a side contralateral signal relating to the frequency-domain signal; transforming an ipsilateral signal, generated by mixing the frontal ipsilateral signal and the side ipsilateral signal, and the side contralateral signal into a time-domain ipsilateral signal and a time-domain contralateral signal, which are time-domain signals, respectively; and generating a binaural signal by mixing the time-domain ipsilateral signal and the time-domain contralateral signal.

Method for generating binaural signals from stereo signals using upmixing binauralization, and apparatus therefor

The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor. To this end, provided are a method for generating a filter for an audio signal, including: receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of an input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining average reverberation time information of a corresponding subband by using reverberation time information extracted from the subband filter coefficients; obtaining at least one coefficient for curve fitting of the obtained average reverberation time information; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; obtaining filter order information for determining a truncation length of the subband filter coefficients, the filter order information being obtained by using the average reverberation time information or the at least one coefficient according to the obtained flag information and the filter order information of at least one subband being different from filter order information of another subband; and truncating the subband filter coefficient by using the obtained filter order information and a parameterization device therefor.

Method for generating filter for audio signal, and parameterization device for same

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.

Method and apparatus for processing multimedia signals

An audio signal processing apparatus is provided. The audio signal processing apparatus includes an input terminal receiving an input audio signal, a processor obtaining a difference between a playback loudness of the input audio signal and a desired loudness thereof and correcting a frequency band spectrum of an output audio signal for each of a plurality of frequency bands based on the difference between the playback loudness and the desired loudness of the input audio signal and a relationship between a loudness and a sound pressure for each of the plurality of frequency bands, and an output terminal outputting the output audio signal. The playback loudness is a loudness of the output audio signal when the input audio signal is output without the correction.

Audio signal processing method and apparatus for frequency spectrum correction

Disclosed is an audio signal processing device. The audio signal processing device includes a receiving end configured to receive a 2-channel stereo signal and a processor configured to process the 2-channel stereo signal. The processor is configured to filter the 2-channel stereo signal using a spatial distortion removal filter and output the filtered 2-channel stereo signal to a speaker including two or more channels. The spatial distortion removal filter is a filter for offsetting distortion that occurs when the output signal is transmitted from the speaker to a listener, and includes an ipsilateral filter applied to an ipsilateral signal of the 2-channel audio signal and a contralateral filter applied to a contralateral signal of the 2-channel audio signal.

Method and device for processing audio signals using 2-channel stereo speaker

The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal. To this end, provided are a method for processing an audio signal, which includes: receiving an input audio signal including a multi-channel signal; receiving truncated subband filter coefficients for filtering the input audio signal, the truncated subband filter coefficients being at least some of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal and the length of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using reverberation time information extracted from the corresponding subband filter coefficients; obtaining vector information indicating the BRIR filter coefficients corresponding to each channel of the input audio signal; and filtering each subband signal of the multi-channel signal by using the truncated subband filter coefficients corresponding to the relevant channel and subband based on the vector information and an apparatus for processing an audio signal by using the same.

Audio signal processing method and apparatus

A method of generating audio loudness performed by an audio loudness generation device may include: receiving loudness information on each of a plurality of audio tracks included in one group; predicting an intermediate loudness distribution, which is a loudness distribution for the one group, on the basis of the loudness information on each of the plurality of audio tracks; and generating an integrated loudness for the one group on the basis of the intermediate loudness distribution.

Method for generating audio loudness metadata and device therefor

Growing community of inventors

Gyeonggi-do, South Korea

Taegyu Lee