Samsung Electronics Co., Ltd.

Industry-University Cooperation Foundation Hanyang University

Ho-Sang Sung

Eun-mi Oh

Ki-hyun Choo

Jong-Hoon Jeong

Eun-Mi Oh

Eun Mi Oh

Jong-youb Ryu

Jung-Hoe Kim

Kyung-hun Jung

Holly Francois

Ki-Hyun Choo

Chang-Yeong Kim

Anton Porov

Byeong-hoon Kwak

Jong-Hyun Kim

Nam-Suk Lee

Anton Victorovich Porov

Kyung-Hun Jung

Sang-Won Kang

Yaxing Li

Betlehem Terence

Chang-Yong Son

Kang-Eun Lee

Jong-Youb Ryu

Byeong-Hoon Kwak

Provided are an audio reconstruction method and device for providing improved sound quality by reconstructing a decoding parameter or an audio signal obtained from a bitstream, by using machine learning. The audio reconstruction method includes obtaining a plurality of decoding parameters of a current frame by decoding a bitstream, determining characteristics of a second parameter included in the plurality of decoding parameters and associated with a first parameter, based on the first parameter included in the plurality of decoding parameters, obtaining a reconstructed second parameter by applying a machine learning model to at least one of the plurality of decoding parameters, the second parameter, and the characteristics of the second parameter, and decoding an audio signal, based on the reconstructed second parameter.

Audio reconstruction method and device which use machine learning

Disclosed is an electronic apparatus. The electronic apparatus includes a storage for storing a plurality of filters trained in a plurality of convolutional neural networks (CNNs) respectively and a processor configured to acquire a first spectrogram corresponding to a damaged audio signal, input the first spectrogram to a CNN corresponding to each frequency band to apply the plurality of filters trained in the plurality of CNNs respectively, acquire a second spectrogram by merging output values of the CNNs to which the plurality of filters are applied, and acquire an audio signal reconstructed based on the second spectrogram.

Electronic device and a controlling method thereof

A voice signal processing method includes acquiring a near-end noise signal and a near-end voice signal by using at least one microphone, acquiring a far-end voice signal according to an incoming call, determining a noise control parameter and a voice signal change parameter based on at least one of information about the near-end voice signal, information about the near-end noise signal, or information about the far-end voice signal, generating an anti-phase signal of the near-end noise signal based on the noise control parameter, changing the far-end voice signal to improve articulation of the far-end voice signal based on information related to at least one of the voice signal change parameter, the near-end noise signal, or the anti-phase signal, and outputting the anti-phase signal and the changed far-end voice signal.

Method and apparatus for processing speech signal adaptive to noise environment

Provided is a method for processing a signal adaptively to a noisy environment, the method including: determining a class related to enhancement of intelligibility, based on a noise spectrum of a reception side or a near-end device and a received voice spectrum; generating guide information for the enhancement of intelligibility based on the noise spectrum, the voice spectrum, and the determined class; and generating a changed voice spectrum by applying the guide information to the voice spectrum.

Signal processing method and device adaptive to noise environment and terminal device employing same

Provided are a method and an apparatus for increasing call quality while reducing consumption of bit rates in immersive multimedia communication. A method of transmitting, by a terminal device, an audio signal to a counterpart terminal device, includes: extracting head tracking information of a user of the counterpart terminal device from a packet received from the counterpart terminal device; predicting a head direction of the user of the counterpart terminal device after a predetermined time period, based on the extracted head tracking information; generating audio output signals by combining audio input signals received via a plurality of microphones, based on the predicted head direction; and transmitting information about the predicted head direction and a sound frame in which the audio output signals are compressed, to the counterpart terminal device.

Method for transmitting audio signal and outputting received audio signal in multimedia communication between terminal devices, and terminal device for performing same

A voice signal processing method according to an embodiment of the present disclosure for overcoming the problem includes: acquiring a real-time near-end noise signal; acquiring a far-end voice signal according to an incoming call; measuring subjective speech quality and perceptual-objective speech quality of test signals generated based on a reference signal and the real-time near-end noise signal; selecting at least one speech quality enhancement method based on the subjective speech quality and the perceptual-objective speech quality, and determining parameters that are to be applied to the selected at least one speech quality enhancement method; and enhancing speech quality of the far-end voice signal by using the selected at least one speech quality enhancement method, based on the determined parameters, wherein the test signals are generated by mixing the acquired real-time near-end noise signal with the reference signal whose speech quality is enhanced by applying a combination of parameter values to speech quality enhancement methods.

Method and apparatus for improving call quality in noise environment

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Seoul, South Korea

Ho-Sang Sung