Abstract:
음성 및 음악 신호를 통합적으로 부호화 및 복호화 하는 장치 및 방법이 제공된다. 상기 부호화 장치(100)는 입력 신호의 특성을 분석하는 입력 신호 분석부(110); 상기 입력 신호를 주파수 영역 신호로 변환하고, 상기 입력 신호가 음성 특성 신호인 경우 상기 입력 신호를 부호화하는 제1 변환 부호화부(120); 상기 입력 신호가 음성 특성 신호인 경우 상기 입력 신호를 선형예측코딩(LPC)에 기반하여 부호화하는 선형예측코딩(LPC) 부호화부; 그리고, 상기 제1 변환 부호화부(120)의 출력 신호 및 상기 LPC 부호화부(130)의 출력 신호를 사용하여 비트스트림을 생성하는 비트스트림 발생부(140)를 포함할 수 있다.
Abstract:
PURPOSE: A method and system for separating a music sound source without using a sound source database are provided to separate a sound source generated using a rhythm musical instrument based on characteristics of the rhythm musical instrument repeated in an aspect of time, and thereby may separate a sound source included in a mixed signal even when a learning database generated using a specific sound source is absent. CONSTITUTION: An NMPCF(Nonnegative Matrix Partial Co-Factorization) analysis unit(130) obtains info nation shared by the plurality of segments in accordance with an NMPCF algorithm. A rhythm instrument signal has frequency characteristics such as a pitch, that may not be easily changed, and may be repeatedly generated, whereby the shared information may correspond to information of a rhythm musical instrument. A target instrument signal separating unit(140) separates a target instrument signal corresponding to a specific sound source from the mixed signal by calculating an inner product between the entity matrices obtained by the NMPCF analysis unit. The target instrument signal is a signal including sounds generated using the rhythm musical instrument. A signal association unit(150) associates the target instrument signals for each of the plurality of segments separated in the target instrument signal separating unit.
Abstract:
PURPOSE: An apparatus and a method for structuring bitstream for an object-based audio service are provided to reduce a degradation in the sound quality occurring due to an excessive volume control by designating an upper bound value and a lower bound value of a reproduction volume in an object-based audio service. CONSTITUTION: A bitstream splitting unit divides a bitstream into a file header, and an audio object frame through sound source splitting(110). A reproduction level information storage unit stores the reconstruction level information for the regeneration of the audio object within the file header(120). A preset storage unit stores the preset information for the regeneration of the audio object within the file header(130).
Abstract:
PURPOSE: A system for separating musical sound source is provided to efficiently divide a specific sound source in a mixed sound source by reorganizing the mixed sound source into the specific sound source and other sound source. CONSTITUTION: A system for separating musical sound source includes a database(110), a time-frequency domain conversion unit(120), an NMPCF(Nonnegative Matrix Partial Co-Factorization) analysis unit(130), a target instrument signal separating unit(140), and a time domain signal conversion unit(150). The database stores information about a solo performance using a predetermined musical instrument, and transmits the information about the solo performance as a type of a predetermined sound source signal(x1). The predetermined sound source has a significantly great amount of data to include various characteristics of the predetermined sound source. In this case, a great amount of database signals may need to be processed for each sound source separation operation.
Abstract:
PURPOSE: A unified speech/audio coder(USAC) processing windows sequence based mode switching is provided to perform encoding or decoding by processing different window sequences for different situation, thereby improving coding efficiency. CONSTITUTION: A mode switching unit performs inter-linear prediction domain mode switching for the sub-frames which constitute a frame of an input signal. An encoding unit applies a window sequence to current encoding subframes, wherein the window sequence is based on a switched LPD(Linear Prediction Domain) mode. In addition, the encoding unit applies a changed window sequence to the current encoding subframe according to the LPD modes of previous and following subframes.
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.