Abstract: An audio encoding method and apparatus for electronic shock protection (ESP) function in an audio player. A method of generating a frame structure for an electronic shock protection function of an audio player, includes: encoding original data of a frame and locating the encoded data in a first area that is a start part of the frame structure; and encoding frame length information indicating a size of the encoded data in the frame, and locating the encoded frame length information in a second area that is an end part of the frame structure.

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: Provided is a touch panel using a zinc oxide (ZnO) nano wire. The touch panel may include a first transparent substrate, a first transparent electrode layer on the first transparent substrate, a light transmissive nano wire layer including a plurality of piezoelectric nano wires that may be arranged on the first transparent electrode layer so as to be perpendicular to the first transparent electrode layer, a second transparent electrode layer on the nano wire layer, and a second transparent substrate on the second transparent electrode layer.

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 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.

Abstract: A decoding method, medium, and device decoding an input signal, including compressed multi-channel signals as a mono or stereo signal, into 2-channel binaural signals. A full band channel level of each channel in the multi-channel system is calculated from channel level differences between the channels, and data of each channel included in the input signal is localized in directions corresponding to the channels based on the calculated full band channel levels of the channels. Accordingly, the input signal can be output as the 2-channel binaural signals by using simple operations without having to reconstruct multi-channel signals from the input signal in a quadrature mirror filter (QMF) domain.

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 device and method for encoding/decoding a speech signal and an audio signal. The device for encoding the speech signal and the audio signal includes a speech encoding unit which speech-encodes an input signal; an speech decoding unit which speech-decodes the speech-encoded signal; and an audio encoding unit which divides a difference signal between the speech-decoded signal and the input signal into a low band and a high band, allocates the number of bits to the divided bands, and audio-encodes the difference signal.

Abstract: A method and apparatus to encode/decode an audio signal, in which a bit rate for each bit plane can be controlled. A method of encoding an audio signal for each of a plurality of bit plane can include dividing the audio signal into a plurality of frequency bands and encoding the bit planes of the frequency bands from a low frequency band to a high frequency band, wherein, in the encoding the bit planes of the frequency bands, the bit planes are encoded from the most significant bit (MSB) to the least significant bit (LSB) within bits allocated for the frequency bands, and when there are allocated bits remaining after the encoding of the currently encoded frequency band, un-encoded bit planes corresponding to the MSB in a frequency band that has the fewest encoded bit planes among frequency bands with a lower frequency than the currently encoded frequency band are encoded using the remaining allocated bits.

Abstract: 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 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.

Abstract: A method and apparatus to scalably encode and/or decode an audio signal includes encoding a specific band signal included in an input signal, encoding a frequency envelope of an excited signal in which the encoded specific band signal is removed from the input signal, encoding a residual signal in which the encoded frequency envelope is removed from the excited signal, and forming a bit-stream by scalably packing the encoded specific band signal, frequency envelop, and residual signal.

Abstract: Provided are a scalable wide-band speech coding/decoding apparatus, method, and medium. An input wide-band speech input signal is first divided into a low-band signal and a high-band signal. The divided low-band signal is then coded using a code excited linear prediction (CELP) method. The divided high-band signal is coded using a harmonic method. A signal representing a difference between a synthetic signal obtained from the low-band and the high band, and a signal input to the low-band and the high-band is then coded using a modified discrete cosine transform (MDCT) method. The coded signal is then multiplexed. The multiplexed signal is then output. Accordingly, high quality speech can be achieved for all layers.

Abstract: An arrangement is provided for compressing microcode ROM (“uROM”) in a processor and for efficiently accessing a compressed “uROM”. A clustering-based approach may be used to effectively compress a uROM. The approach groups similar columns of microcode into different clusters and identifies unique patterns within each cluster. Only unique patterns identified in each cluster are stored in a pattern storage. Indices, which help map an address of a microcode word (“uOP”) to be fetched from a uROM to unique patterns required for the uOP, may be stored in an index storage. Typically it takes a longer time to fetch a uOP from a compressed uROM than from an uncompressed uROM. The compressed uROM may be so designed that the process of fetching a uOP (or uOPs) from a compressed uROM may be fully-pipelined to reduce the access latency.

Abstract: A method and apparatus to quantize/dequantize frequency amplitude data and a method and apparatus to audio encode/decode using the method and apparatus to quantize/dequantize the frequency amplitude data. The method includes calculating and quantizing power of frequency amplitudes for each of a plurality of bands constituting an audio frame, normalizing frequency amplitude data for each of the bands using the quantized power, and quantizing a first one of even-numbered or odd-numbered data among the normalized frequency amplitude data.

Abstract: 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.

Abstract: In one embodiment, the present invention includes a method to compress data stored in a memory to reduce size and power consumption. The method includes segmenting each word of a code portion into multiple fields, forming tables having unique entries for each of the fields, and assigning a pointer to each of the unique entries in each of the tables. Other embodiments are described and claimed.

Abstract: 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.

Abstract: 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.

Abstract: A method and an apparatus for encoding/decoding a digital signal are provided. First, a digital input signal is transformed into samples to remove redundant information among signals. Then, a lookup table corresponding to a characteristic of the input signal is selected among a plurality of lookup tables that indicate different numbers of bits allocated for each quantization unit depending on different characteristics of input signals, and the number of bits allocated for each quantization unit is acquired from the selected lookup table. Next, a distribution of samples within each quantization unit is divided into a predetermined number of sections, and the samples are linearly quantized using the allocated number of bits on a section-by-section basis. Thereafter, a bitstream comprised of frames is produced from the quantized samples and predetermined side information so that information about a frame length is stored in the end of frame.

Abstract: An audio encoding method and apparatus for electronic shock protection (ESP) function in an audio player. A method of generating a frame structure for an electronic shock protection function of an audio player, includes: encoding original data of a frame and locating the encoded data in a first area that is a start part of the frame structure; and encoding frame length information indicating a size of the encoded data in the frame, and locating the encoded frame length information in a second area that is an end part of the frame structure.