Abstract: 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.

Abstract: A method of encoding/decoding a digital signal using linear quantization by sections, and an apparatus for the same are provided. The method of encoding includes: converting a digital input signal, and removing redundant information from the digital signal; allocating a number of bits allocated to each predetermined quantized unit considering the importance of the digital signal; dividing the distribution of signal values into predetermined sections based on the predetermined quantized units, and linear quantizing data converted pin the operation of converting the digital input signal by sections; and generating a bit stream from the linear quantized data and predetermined side information. Therefore, a sound quality is improved compared to a sound quality produced by conventional linear quantizing devices and a complexity of a non-linear quantizing device is reduced.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A method of and an apparatus for encoding/decoding an MPEG-4 bit sliced arithmetic coding (BSAC) audio bitstream having ancillary information. A time domain audio signal is converted to a frequency domain audio signal and quantized. A number of data bits is counted and a number of available bits per layer is obtained. The number of available bits per layer is modified considering the size of ancillary information. Actual audio data is encoded in units of layers and ancillary information is embedded in the encoded bitstream. A header is decoded and a layer structure of an audio bitstream is calculated to determine the size of the ancillary information as a difference between a size of data up to a top layer and a size of a frame. The ancillary information is extracted to improve meta data and sound quality of audio contents.

Abstract: According to one embodiment a computer system is disclosed. The computer system includes a microprocessor and a chipset coupled to the microprocessor. The microprocessor removes stale branch instructions prior to the execution of a first cache line by finding existing branch prediction data for the first cache line.

Abstract: A predictor of consecutive useless accesses, wherein consecutive useless accesses to a logic unit are counted and a next useless access is predicted to be within a plurality of ranges. Each of the plurality of ranges has a corresponding confidence predictor to track and provide a confidence level of whether a next access to the logic unit will be useless.