摘要:
A lossless audio encoding/decoding method, medium, and apparatus. The lossless audio encoding method includes converting an audio signal in a time domain into an audio spectral signal with an integer in a frequency domain, mapping the audio spectral signal in the frequency domain to a bit plane signal according to its frequency, and losslessly encoding binary samples of bit planes using a probability model determined according to a predetermined context. The lossless audio decoding method includes extracting a predetermined lossy bitstream and an error bitstream from error data by demultiplexing an audio bitstream, the error data corresponding to a difference between lossy encoded audio data and an audio spectral signal with an integer in a frequency domain, lossy decoding the extracted encoded lossy bitstream, losslessly decoding the extracted error bitstream, and restoring the original audio frequency spectral signal using the decoded lossy bitstream and error bitstream
摘要:
A lossless audio coding and/or decoding method and apparatus are provided. The coding method includes: mapping the audio signal in the frequency domain having an integer value into a bit-plane signal with respect to the frequency; obtaining a most significant bit and a Golomb parameter for each bit-plane; selecting a binary sample on a bit-plane to be coded in the order from the most significant bit to the least significant bit and from a lower frequency component to a higher frequency component; calculating the context of the selected binary sample by using significances of already coded bit-planes for each of a plurality of frequency lines existing in the vicinity of a frequency line to which the selected binary sample belongs; selecting a probability model by using the obtained Golomb parameter and the calculated contexts; and lossless-coding the binary sample by using the selected probability model. According to the method and apparatus, a compression ratio better than that of the bit-plane Golomb code (BPGC) is provided through context-based coding method having optimal performance.
摘要:
A lossless audio coding and/or decoding method and apparatus are provided. The coding method includes: mapping the audio signal in the frequency domain having an integer value into a bit-plane signal with respect to the frequency; obtaining a most significant bit and a Golomb parameter for each bit-plane; selecting a binary sample on a bit-plane to be coded in the order from the most significant bit to the least significant bit and from a lower frequency component to a higher frequency component; calculating the context of the selected binary sample by using significances of already coded bit-planes for each of a plurality of frequency lines existing in the vicinity of a frequency line to which the selected binary sample belongs; selecting a probability model by using the obtained Golomb parameter and the calculated contexts; and lossless-coding the binary sample by using the selected probability model. According to the method and apparatus, a compression ratio better than that of the bit-plane Golomb code (BPGC) is provided through context-based coding method having optimal performance.
摘要:
A method, medium, and apparatus for converting compressed audio data, including decoding compressed audio input data, in accordance with a corresponding compression format, coding a result of the decoding, in accordance with a predetermined compression format, and combining a result of the coding with the side information to generate audio output data to be compressed according to the predetermined compression format.
摘要:
A lossless audio encoding/decoding method, medium, and apparatus. The lossless audio encoding method includes converting an audio signal in a time domain into an audio spectral signal with an integer in a frequency domain, mapping the audio spectral signal in the frequency domain to a bit plane signal according to its frequency, and losslessly encoding binary samples of bit planes using a probability model determined according to a predetermined context. The lossless audio decoding method includes extracting a predetermined lossy bitstream and an error bitstream from error data by demultiplexing an audio bitstream, the error data corresponding to a difference between lossy encoded audio data and an audio spectral signal with an integer in a frequency domain, lossy decoding the extracted encoded lossy bitstream, losslessly decoding the extracted error bitstream, and restoring the original audio frequency spectral signal using the decoded lossy bitstream and error bitstream.
摘要:
A method, medium, and apparatus for converting compressed audio data, including decoding compressed audio input data, in accordance with a corresponding compression format, coding a result of the decoding, in accordance with a predetermined compression format, and combining a result of the coding with the side information to generate audio output data to be compressed according to the predetermined compression format.
摘要:
A multichannel audio data encoding and/or decoding method and apparatus. The encoding method includes: encoding mono and/or stereo audio data; and encoding extended multichannel audio data other than the mono and/or stereo audio data. The decoding method includes: decoding mono and/or stereo audio data; examining whether there is extended multichannel audio data to be decoded other than the mono and/or stereo audio data; and when there is extended data to be decoded, decoding the extended multichannel audio data.
摘要:
A digital signal encoding method and apparatus using a plurality of lookup tables. The method includes: preparing a plurality of lookup tables storing numbers of allocated bits for encoding frequency bands of an input signal according to a characteristic of the input signal in a predetermined number of addresses; dividing an input signal in the time domain into signals in predetermined frequency bands; calculating address values of the frequency bands; selecting one of the plurality of lookup tables according to the characteristic of the input signal; extracting numbers of allocated bits of addresses having the calculated address values from the selected lookup table with respect to the frequency bands and allocating the numbers of bits to the frequency bands; and generating a bitstream by quantizing the input signal according to the numbers of allocated bits. Bit amount control suitable for a characteristic of an input signal can be performed by extracting numbers of allocated bits of frequency bands from an optimal lookup table selected according to the characteristic of the input signal. Also, an additional computational time can be reduced by using each occupancy rate per frequency band equal to each address of the lookup table as the characteristic of the input signal.
摘要:
An apparatus and method encode audio data, and an apparatus and method decode encoded audio data. An audio data encoding apparatus includes: a scalable encoding unit dividing audio data into a plurality of layers, representing the audio data in predetermined numbers of bits in each of the plurality of layers, and encoding a lower layer prior to encoding an upper layer and an upper bit of each layer prior to encoding a lower bit of each layer; an SBR encoding unit generating spectral band replication (SBR) data that has information with respect to audio data in a frequency band of frequencies equal to or greater than a predetermined frequency among the audio data to be encoded, and encoding the SBR data; and a bitstream production unit generating a bitstream using the encoded SBR data and the encoded audio data corresponding to a predetermined bitrate.
摘要:
A digital signal encoding method and apparatus using a plurality of lookup tables. The method includes: preparing a plurality of lookup tables storing numbers of allocated bits for encoding frequency bands of an input signal according to a characteristic of the input signal in a predetermined number of addresses; dividing an input signal in the time domain into signals in predetermined frequency bands; calculating address values of the frequency bands; selecting one of the plurality of lookup tables according to the characteristic of the input signal; extracting numbers of allocated bits of addresses having the calculated address values from the selected lookup table with respect to the frequency bands and allocating the numbers of bits to the frequency bands; and generating a bitstream by quantizing the input signal according to the numbers of allocated bits. Bit amount control suitable for a characteristic of an input signal can be performed by extracting numbers of allocated bits of frequency bands from an optimal lookup table selected according to the characteristic of the input signal. Also, an additional computational time can be reduced by using each occupancy rate per frequency band equal to each address of the lookup table as the characteristic of the input signal.