Electronics and Telecommunications Research Institute

Dolby Laboratories Licensing Corporation

Broadcom Corporation

Yonsei University

Hong-Goo Kang

Kyung-Tae Kim

Dae-Hwan Hwang

Young-Cheol Park

Ki-Seung Lee

Dae-Hee Youn

Ho-Sang Sung

Sung-Kyo Jung

Jeungook Song

Chang Heon Lee

Hyen-O Oh

Juin-Hwey Chen

Jihyun Lee

Tae Jin Lee

Seung Kwon Beack

Jes Thyssen

Robert W Zopf

Jongmo Sung

Woo-taek Lim

Inseon Jang

Hyungseob Lim

An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.

Alias cancelling during audio coding mode transitions

An audio signal encoding and decoding method using a neural network model, and an encoder and decoder for performing the same are disclosed. A method of encoding an audio signal using a neural network model, the method may include identifying an input signal, generating a quantized latent vector by inputting the input signal into a neural network model encoding the input signal, and generating a bitstream corresponding to the quantized latent vector, wherein the neural network model may include i) a feature extraction layer generating a latent vector by extracting a feature of the input signal, ii) a plurality of downsampling blocks downsampling the latent vector, and iii) a plurality of quantization blocks performing quantization of a downsampled latent vector.

Audio signal encoding and decoding method using a neural network model to generate a quantized latent vector, and encoder and decoder for performing the same

A hybrid instantaneous/differential encoding technique is described herein that may be used to reduce the bit rate required to encode a pitch period associated with a segment of a speech signal in a manner that will result in relatively little or no degradation of a decoded speech signal generated using the encoded pitch period. The hybrid instantaneous/differential encoding technique is advantageously applicable to any speech codec that encodes a pitch period associated with a segment of a speech signal.

Hybrid instantaneous/differential pitch period coding

Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.

Dynamic time scale modification for reduced bit rate audio coding

A system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, encoding a difference between a baseband speech signal and a standard baseband between a synthesized standard baseband signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

System and method for transmitting and receiving wideband speech signals with a synthesized signal

The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Transmitter and receiver for speech coding and decoding by using additional bit allocation method

The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

System and method for transmitting and receiving wideband speech signals

Growing community of inventors

Seoul, South Korea

Hong-Goo Kang