Abstract:
PURPOSE: An apparatus and a method for encoding and decoding an integrated voice and music signal are provided to effectively select internal modules according to a characteristic of an input signal, thereby providing excellent sound quality to both a voice signal and a music signal at various bit rates. CONSTITUTION: An input signal analysis unit(110) analyzes a characteristic of an input signal. When the input signal is a music characteristic signal, the first conversion encoding unit(120) converts the input signal into a frequency domain to encode. When the input signal is a voice characteristic signal, an LPC(Linear Predictive Coding) encoder encodes the input signal based on LPC. A bit stream generator(140) uses output signals of the first conversion encoding unit and the LPC encoder to generate a bit stream.
Abstract:
PURPOSE: An apparatus for determining a signal state of an audio signal is provided to suitably select a voice encoder based on LPC(Linear Predictive Coding) and an audio encoder based on conversion according to a characteristic of an input signal. CONSTITUTION: A signal state observer(101) classifies characteristics of an input signal to output each state observation probability. A state chain unit(102) outputs a state identifier of a frame of the input signal based on the state observation probability. An encoder encoding a frame of the input signal is determined according to the state identifier. The signal state observer comprises a characteristic extracting unit, an entropy base determining tree and a silence state determining unit.
Abstract:
A method and an apparatus for solving the permutation ambiguity of a separation coefficient of a multi-channel mixed signal are provided to prevent the mixing of different sound source signals, thereby accurately separating an original signal from the mixed signal. An ICA(Independent Component Analysis) input matrix is created in a row vector type of a back-mixing matrix corresponding to a separation coefficient(401). A base vector matrix and a deflection coefficient matrix are extracted using the created ICA method(402). A base vector index value regarding a column vector of a mixing vector is selected by comparing the extracted base vector matrix and deflection coefficient matrix with each other(403). A corresponding clustering including the column vector of the mixing vector is determined based on the selected base vector index value(404).
Abstract:
모드 스위칭에 기초하여 윈도우 시퀀스를 처리하는 통합 음성/오디오 부/복호화기가 개시된다. 통합 음성/오디오 부/복호화기는 모드 스위칭이 발생하는 경우, 폴딩 포인트를 기준으로 프레임 간 오버랩을 수행하여 부호화하거나 또는 복호화할 수 있다. 통합 음성/오디오 부/복호화기는 부호화 또는 복호화를 수행하기 위해 상황에 따라 다른 윈도우 시퀀스를 처리함으로써, 코딩 성능을 향상시킬 수 있다. USAC(Unified Speech/Audio Coder), 윈도우 시퀀스, FD, MDCT, LPD
Abstract:
PURPOSE: An integrated voice/audio coding/decoding device for processing a window sequence based on mode switching is provided to differently perform a coding method corresponding to a property of an input signal, thereby maximizing coding performance and sound quality. CONSTITUTION: An integrated voice/audio coding/decoding device performs LPD(Linear Prediction Domain) mode switching for a sub-frame which comprises a frame of an input signal. The integrated voice/audio coding/decoding device codes the input signal by applying a window based on an LPD mode to a current sub-frame. The integrated voice/audio coding/decoding device applies the window to the current sub-frame. The window is changed corresponding to an LPD mode of a previous sub-frame and an LPD mode of a next sub-frame.