摘要:
Provided is a filtering apparatus, method, and medium for a multi-format codec, in which a blocking artifact of decoded video data is removed. The filtering apparatus includes a compression format detection unit, a block strength determination unit, a table determination unit, and a first filtering unit. The compression format detection unit detects a video compression format of the decoded video data. The block strength determination unit determines a block strength indicating a filtering strength with respect to the decoded video data according to the detected video compression format. The table determination unit determines a filtering table differently according to the block strength when the block strength is greater than a predetermined threshold. The first filtering unit performs filtering on the decoded video data in units of a block using the determined filtering table.
摘要:
Provided is a scalable encoding method, apparatus, and medium. The method includes: encoding a base layer and encoding a first enhancement layer and a second enhancement layer in a frame having the base layer; and generating an encoded frame by synthesizing the encoded results. Accordingly, only if the loss of the encoding frame is not as great as the encoded first enhancement layer is damaged, a case where speech restoration with respect to partial frequency bands must be given up does not occur. Furthermore, since an encoder divides the second enhancement layer into a plurality of layers in a horizontal or vertical direction, considering a distribution pattern of data belonging to the second enhancement layer and first encodes a layer in which lots of data are distributed among the divided layers, loss of audio information can be minimized even if a portion of the encoded second enhancement layer is damaged.
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 display driving apparatus and method using a display device, such as a liquid crystal display device, and a medium for implementing a method of display driving method are provided. The display driving apparatus includes: an encoding unit which compresses and encodes image data to be displayed by the display device; a memory which stores the encoded image data; a memory control unit which stores the encoded image data in the memory and reads the encoded image data from the memory; and a decoding unit which decodes the encoded image data read out from the memory to restore the image data. Input image data is compressed and encoded in units of blocks, and the encoded image data is stored in a memory. Thereafter, the encoded image data stored in the memory is decoded, and the decoded result is output to a display device. Thus, it is possible to reduce the size of a memory included in a display driving apparatus without deteriorating the quality of an image displayed.
摘要:
An apparatus for and a method of processing a multi-channel audio signal using space information. The apparatus includes: a main coding unit down mixing a multi-channel audio signal by applying space information to surround components included in the multi-channel audio signal, generating side information using the multi-channel audio signal or a stereo signal of a down-mixed result, coding the stereo signal and the side information, and transmitting the coded result as a coding signal; and a main decoding unit receiving the coding signal, decoding the stereo signal and the side information using the received coding signal, up mixing the decoded stereo signal using the decoded side information, and restoring the multi-channel audio signal.
摘要:
Provided are an audio encoding method and apparatus capable of fast bit rate control. The audio encoding method includes: converting audio sampling data into frequency domain data; adjusting a scalefactor value in each predetermined frequency band based on an available bits and allowed distortion of a psychoacoustic model to allocate a number of necessary bits to the frequency domain data and quantize the frequency domain data; and generating a bit stream based on the quantized data. The quantizing of the frequency domain data includes: obtaining the available bits for the frequency domain data; obtaining the common scalefactor value satisfying that the used bits is not larger than the available bits, using a difference the available bits and the used bits to quantize the audio data; calculating quantization noise in the each predetermined quantization band; and adjusting a scalefactor value of a quantization band in which the quantization noise exceeds the allowed distortion of the psychoacoustic model to quantize the 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:
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 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.
摘要:
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.