摘要:
Provided is an audio encoding method. The audio encoding method includes: acquiring envelopes based on a predetermined sub-band for an audio spectrum; quantizing the envelopes based on the predetermined sub-band; and obtaining a difference value between quantized envelopes for adjacent sub-bands and lossless encoding a difference value of a current sub-band by using a difference value of a previous sub-band as a context. Accordingly, the number of bits required to encode envelope information of an audio spectrum may be reduced in a limited bit range, thereby increasing the number of bits required to encode an actual spectral component.
摘要:
Provided is an audio encoding method. The audio encoding method includes: acquiring envelopes based on a predetermined sub-band for an audio spectrum; quantizing the envelopes based on the predetermined sub-band; and obtaining a difference value between quantized envelopes for adjacent sub-bands and lossless encoding a difference value of a current sub-band by using a difference value of a previous sub-band as a context. Accordingly, the number of bits required to encode envelope information of an audio spectrum may be reduced in a limited bit range, thereby increasing the number of bits required to encode an actual spectral component.
摘要:
A multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.
摘要:
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.
摘要:
A lossless coding and/or decoding apparatus and method. The lossless coding apparatus may read a probability model corresponding to each of a plurality of context groups. Here, the probability model stored in a memory may be generated by grouping a context. The lossless coding apparatus may code a symbol using the probability model and generate a bitstream. The lossless coding apparatus may enhance coding efficiency and reduce an amount of space utilized by the memory.
摘要:
Provided is an audio signal encoding method including transforming an input signal from a time domain to a time/frequency domain using a first transformation method, extracting a stereo parameter from a signal of the time/frequency domain, encoding the stereo parameter, and down-mixing the signal of the time/frequency domain, transforming each of sub-bands of the down-mixed signal to a frequency domain by using a second transformation method, and encoding the signal of the frequency domain in the frequency domain.
摘要:
Provided is an audio signal encoding method including transforming an input signal from a time domain to a time/frequency domain using a first transformation method, extracting a stereo parameter from a signal of the time/frequency domain, encoding the stereo parameter, and down-mixing the signal of the time/frequency domain, transforming each of sub-bands of the down-mixed signal to a frequency domain by using a second transformation method, and encoding the signal of the frequency domain in the frequency domain.
摘要:
A lossless coding and/or decoding apparatus and method. The lossless coding apparatus may read a probability model corresponding to each of a plurality of context groups. Here, the probability model stored in a memory may be generated by grouping a context. The lossless coding apparatus may code a symbol using the probability model and generate a bitstream. The lossless coding apparatus may enhance coding efficiency and reduce an amount of space utilized by the memory.
摘要:
A low-bitrate encoding system includes: a time-frequency transform unit transforming an input time-domain audio signal into a frequency-domain audio signal; a frequency component processor unit decimating frequency components in the frequency-domain audio signal; a psychoacoustic model unit modeling the received time-domain audio signal on the basis of human auditory characteristics, and calculating encoding bit allocation information; a quantizer unit quantizing the frequency-domain audio signal input from the frequency component processor unit to have a bitrate based on the encoding bit allocation information input from the psychoacoustic model unit; and a lossless encoder unit encoding the quantized audio signal losslessly, and outputting the encoded audio signal in a bitstream format. Using the low-bitrate encoding system, it is possible to effectively compress data at a low bitrate, and thus to provide a high quality audio signal.
摘要:
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.