Abstract:
A method and an apparatus for encoding and decoding a signal are provided to restore an audio signal and a voice signal into high quality signals by effectively encoding and decoding the audio signal and the voice signal with a small number of bits. Frequency or time resolution for each frequency band is determined by applying a psycho acoustic model. An input signal is converted according to the frequency or time resolution determined by each frequency band. The determined frequency or time resolution is compared with a predetermined value per each frequency band and selectively encoded by a predetermined method. The converted signal is quantized and encoded.
Abstract:
A method and an apparatus for encoding and decoding audio signals are provided to maximize coding efficiency since sound quality of audio signal is not lowered while using less bit in encoding and decoding. An encoding method of audio signal includes a step for detecting and encoding frequency components in an input signal according to set standards(2010,2015), and a step for calculating and encoding an energy value about the input signal according to band(2036,2037). A decoding method of audio signal includes a step for decoding frequency components, a step for decoding energy value of the signal prepared for each band, a step for calculating the energy value of the signal generated in each band in consideration of the energy value of the decoded frequency components based on the decoded energy value, a step for producing a signal having the calculated energy value according to band and a step for synthesizing the frequency components and the generated signals.
Abstract:
A method for encoding an audio/speech signal in the time domain is provided to detect an envelop in a linear prediction analysis, apply an adjustable codebook and a fixed codebook adjustably, thereby improving compressing efficiency and audio quality by reflecting properties of an input signal when the audio/speech signal is encoded. An envelop of an input signal is detected according to the location of attack of the input signal. An adaptive codebook is searched according to the controlled resolution of parameter based on information about the attack of the input signal and the residual signal is encoded. A fixed codebook is searched based on the index controlled according to the location of the attack of the input signal and an excitation signal is encoded(33). In this way, the property of the input signal is reflected so that the audio/speech signal can be encoded effectively.
Abstract:
본 발명은 계층적으로 오디오 데이터를 부호화하는 방법 및 장치, 오디오 데이터를 복호화하는 방법 및 장치에 관한 것으로, 오디오 데이터를 계층적으로 부호화하고, 적어도 하나 이상의 부호화 기법에 따라 오디오 데이터의 적어도 하나 이상의 확장 데이터를 부호화하고, 또한 이와 같은 방식으로 복호화함으로서 FGS를 그대로 지원하면서도 오디오 데이터의 확장성을 거의 무한하게 하였다.
Abstract:
A coding apparatus and a method for adaptively applying a window size are provided to improve a compression efficiency and sound quality and to apply a window size by a kind of subbands according to signal characteristics. A band dividing unit(100) divides an input signal into a plurality of subbands. A window size determining unit(110) determines a window size to be applied to each divided subband. A conversion unit(120) converts a signal of each subband from a time domain to a frequency domain with the determined window size. A quantization unit(130) quantizes the converted signals. The window size determining unit determines the window size by using an energy value of each divided subband or by using the degree in which a signal of each divided subband changes.
Abstract:
A method and an apparatus for concealing an error of an audio signal, and a method and an apparatus for decoding the audio signal by using the same are provided to apply a repeat method with an interpolation method of a frequency region, and to recover a frame having an error, thereby precluding sound quality from being lowered due to the modulation noise. When an error occurs in a present frame, one of a frequency section method and a time section method is selected to conceal the error. One of a repeat method and an interpolation method is selected to conceal the error by a pre-set standard when the frequency section method is selected(170,180). The error is concealed by the selected method. One of the frequency section method and the time section method is selected to conceal the error based on whether errors of previous and next frames occur and a window type.
Abstract:
A method and an apparatus for encoding and decoding an audio/speech signal are provided to encode and decode all of the speech signal, the audio signal, and a signal in which the speech signal and the audio signal are mixed efficiently. A domain converting unit(400) converts an input signal from a time domain to a frequency domain by the first conversion type and the second conversion type. A frequency domain encoding unit(430) encodes the signal converted by the first conversion type in the frequency domain by using the signal converted by the second conversion type.
Abstract:
A method and an apparatus for encoding and decoding surround expansion data are provided to encode or decode additional information, capable of up-mixing an audio signal as a surround signal, as the expansion data. An audio data decoding unit(705) decodes audio data hierarchically encoded. A termination code detecting unit(720) detects an identification code indicating that a payload of the audio data is terminated. A start code detecting unit(725) detects an identification code indicating that the payload of expansion data is started. An expansion type detecting unit(730) detects a type of the expansion data. A judging unit judges whether or not the detected type indicates additional information for decoding the audio data by a surround type. If the detected type indicates the additional information, an expansion data decoding unit(735) decodes expansion data corresponding to the additional information.
Abstract:
An encoding method for supporting scalability for the extension of audio data, an apparatus therefor, a decoding method therefor, and an apparatus therefor are provided to reinforce the scalability of audio data and offer backward compatibility capable of supporting an existing BSAC(Bit Sliced Arithmetic Coding) method. The first to the Nth BASC units(62-65) encode audio data and extension data of the audio data. A bit stream processing/transmitting unit(35) groups payloads which are results encoded by the first to the Nth BASC units(62-65) and interleaves the grouped payloads. The bit stream processing/transmitting unit(35) clears a partial group among the interleaved payloads and transmits the other groups.
Abstract:
A method and an apparatus for generating a stereo signal are provided to reduce a system complexity for generating a 3D signal by using a surround data stream by applying an HRTF(Head Related Transfer Function) on a QMF domain. A demultiplexer(300) receives a surround data stream, which contains a down-mixed signal and a spatial parameter, from an encoder, and outputs a demultiplexed result. A domain converter(310) converts the down-mixed signal from the demultiplexer to a QMF(Quadrature Mirror Filter) domain signal. An up-mixer(320) receives the down-mixed signal from the domain converter, decodes the received signal, and up-mixes the decoded result to a multi-channel signal. A stereo signal generator(330) generates a 3D stereo signal on the QMF domain from the up-mixed multi-channel signal. A domain inverse-converter(340) inverse-converts the 3D stereo signal from a QMF domain to a time domain and outputs L-channel and R-channel signals.