Samsung Electronics Co., Ltd.

Xvd Corporation

Alaris Inc.

Gt Technology

Other

Boris D Kudryashov

Victor D Kolesnik

Konstantin Sergeevich Osipov

Irina E Bocharova

Sergey Petrov

Junghoe Kim

Eunmi Oh

Eun-mi Oh

Ki-hyun Choo

Jung-hoe Kim

Boris Trojanovsky

Ki Hyun Choo

Andrey Marsavin

Boris K Trojanovsky

Victor Yu Krachkovsky

Andrey N Trofimov

Andrey Trofimov

Victor Simileysky

Eugene Ovsyannikov

Boris Troyanovsky

Sergei I Kovalov

Evgeny Ovsyannikov

Eugeny P Ovsjannikov

Eun Mi Oh

Mi Young Kim

Jung-Hoe Kim

Felix Aleksandrovich Taubin

Andrei Trofimov

Felix A Taubin

Kirill Yurkov

Eugeny Ovsjannikov

Andrei N Trofimov

Olga Prokofyeva

Andrey Malkov

Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal.

Method and apparatus for generating a stereo signal from a down-mixed mono signal

An apparatus and method of low bit rate encoding and reproducing. The method includes transforming input audio signals in a time domain into spectral signals in a frequency domain, extracting important-spectrum components from the spectral signals in the frequency domain, and quantizing the important-spectrum components, extracting residual-spectrum components other than the important-spectrum components from the spectral signals in the frequency domain, and calculating and quantizing a noise level of the residual-spectrum components, and encoding the quantized important-spectrum components and the quantized noise level losslessly, and outputting encoded bitstreams.

Method and apparatus for low bit rate encoding and decoding

A method and apparatus of effectively encoding and decoding a high-frequency signal of a multi-channel audio are provided. A multi-channel audio decoding apparatus may down-mix a multi-channel audio input signal, expand a number of channels of the down-mixed signal, select at least one of the expanded channel signal, extract a parameter indicating a characteristic relation between the selected signal and the multi-channel audio input signal, and encode the down-mixed signal and the extracted parameter.

Apparatus and method for bandwidth extension for multi-channel audio

Method and apparatus for encoding and decoding stereo signal and multi-channel signal

An method and apparatus to extract an audio signal having an important spectral component (ISC) and a low bit-rate audio signal coding/decoding method using the method and apparatus to extract the ISC. The method of extracting the ISC includes calculating perceptual importance including an SMR (signal-to-mask ratio) value of transformed spectral audio signals by using a psychoacoustic model, selecting spectral signals having a masking threshold value smaller than that of the spectral audio signals using the SMR value as first ISCs, and extracting a spectral peak from the audio signals selected as the ISCs according to a predetermined weighting factor to select second ISCs. Accordingly, the perceptual important spectral components can be efficiently coded so as to obtain high sound quality at a low bit-rate. In addition, it is possible to extract the perceptual important spectral component by using the psychoacoustic model, to perform coding without phase information, and to efficiently represent a spectral signal at a low bit-rate. In addition, the methods and apparatus can be employed in all the applications requiring a low bit-rate audio coding scheme and in a next generation audio scheme.

Method and apparatus to extract important spectral component from audio signal and low bit-rate audio signal coding and/or decoding method and apparatus using the same

A method of encoding a low bit-rate audio signal includes quantizing and encoding a plurality of low frequency sub-bands of an audio signal in a frequency domain, generating a codebook of codevectors using sub-bands of the audio signal spectrum, detecting an envelope of another frequency sub-band of the audio signal and quantizing and losslessly-encoding the detected envelope, selecting a codevector most similar to the higher frequency sub-band spectrum from the generated codebook's codevectors and determining its codebook codevector index, and generating a bit stream. Decoding the low bit-rate audio signal includes restoring and dividing a bit stream into a plurality of first frequency sub-bands and at least one second frequency sub-band and inversely quantizing the first frequency sub-bands in the bit stream, restoring codebook codevector index information and envelope information for the second frequency sub-band, generating a codebook of codevectors using the inversely quantized first frequency sub-bands, and restoring the second frequency sub-band using the restored codevector index information and the envelope information.

Method and apparatus to encode/decode low bit-rate audio signal by approximiating high frequency envelope with strongly correlated low frequency codevectors

A method and apparatus for transforming an audio signal, a method and apparatus for adaptively encoding an audio signal, a method and apparatus for inversely transforming an audio signal, and a method and apparatus for adaptively decoding an audio signal. The method of transforming an audio signal includes determining a transform unit into which the audio signal in a time domain is to be transformed into an audio signal in a frequency domain, and transforming the audio signal into an audio signal in the frequency domain according to the determined transform units using a window coefficient other than 0. Accordingly, it is possible to minimize distortion of the audio signal when encoding the audio signal even at a high bit rate while increasing efficiency of compression.

Method and apparatus for non-overlapped transforming of an audio signal, method and apparatus for adaptively encoding audio signal with the transforming, method and apparatus for inverse non-overlapped transforming of an audio signal, and method and apparatus for adaptively decoding audio signal with the inverse transforming

An apparatus and a method of lossless coding and decoding are provided. The apparatus to perform lossless coding may selectively perform an arithmetic coding scheme or a Huffman coding scheme with respect to a symbol. The apparatus to perform lossless coding may generate a bitstream including a first coding bit, generated according to the Huffman coding scheme. Such bitstream may include a reserved bit for the arithmetic coding scheme.

Apparatus and method of lossless coding and decoding

A method and apparatus to scalably encode or decode an audio signal or a video signal. A signal distribution is considered after binarization is performed based on a tree structure. Thus, a decoding end can minimize generation of quantization errors and restore an original signal. In addition, only frequency components perceptually important in terms of human hearing are set to an upper layer and encoded without decreasing a bandwidth which is restored from an upper layer. Therefore, sound quality can be increased by decreasing quantization noise while maintaining a relatively large frequency bandwidth, and the degradation of a compression performance is prevented.

Scalable encoding and/or decoding method and apparatus

A method and apparatus for audio compression receives an audio signal. Transform coding is applied to the audio signal to generate a sequence of transform frequency coefficients. The sequence of transform frequency coefficients is partitioned into a plurality of non-uniform width frequency ranges and then zero value frequency coefficients are inserted at the boundaries of the non-uniform width frequency ranges. As a result, certain of the transform frequency coefficients that represent high frequencies are dropped.

Method and apparatus for audio compression

A method and apparatus for compressing video signals. The video compression may include row by row horizontal subband filtering of video data representing part of an image, and row by row concurrent vertical high and vertical low pass subband filtering of the results of the horizontal subband filtering. Less than the vertical filter length of rows of results of the horizontal subband filtering may be buffered at a time. The vertical high and vertical low pass subband filtering may include calculating a first result row by adding a first row of a recursion buffer to the results of multiplying a row of results from the horizontal subband filtering and a first vertical filter coefficient, calculating a second result row by adding the first row of the recursion buffer to the result of multiplying the row of results of the horizontal subband filtering and a second vertical filter coefficient, overwriting the first row of the recursion buffer with the second result row, and transmitting the first result row to be encoded.

Apparatus and method for memory saving wavelet based video coding

According to one embodiment of the invention, a multistage vector list quantizer comprises a first stage quantizer to select candidate first stage codewords from a plurality of first stage codewords, a reference table memory storing a set of second stage codewords for each first stage codeword, and a second stage codebook constructor to generate a reduced complexity second stage codebook that is the union of sets corresponding to the candidate first stage codewords selected by the first stage quantizer.

Vector quantization with a non-structured codebook for audio compression

Growing community of inventors

St. Petersburg, Russia

Boris D Kudryashov