Abstract: A method and apparatus to convert a linear predictive coding (LPC) coefficient into a coefficient having order characteristics, such as a line spectrum frequency (LSF), and to vector quantize the coefficient having the order characteristics when a speech signal is encoded. The method and apparatus split the vector of the coefficient having the order characteristics into a plurality of subvectors, select a codebook in which an available bit is variably allocated to each subvector according to distribution of elements of each subvector, and quantize each subvector according to the selected codebook. The method and apparatus use normalized codebooks.

Abstract: Provided are a classifying method and apparatus for an audio signal, and an encoding/decoding method and apparatus for an audio signal using the classifying method and apparatus. In the classification method, an audio signal is classified by adaptively adjusting a classification threshold for a frame of the audio signal that is to be classified according to a long-term feature of the audio signal, thereby improving a hit rate of signal classification, suppressing frequent mode switching per frame, improving noise tolerance, and providing smooth reconstruction of the audio signal.

Abstract: A method and apparatus to determine an encoding mode of an audio signal, and a method and apparatus to encode an audio signal according to the encoding mode. In the encoding mode determination method, a mode determination threshold for the current frame that is subject to encoding mode determination is adaptively adjusted according to a long-term feature of the audio signal for a frame (the current frame) that is subject to encoding mode determination, thereby improving the hit rate of encoding mode determination and signal classification, suppressing frequent oscillation of an encoding mode in frame units, improving noise tolerance, and improving smoothness of a reconstructed audio signal.

Abstract: A high-speed search method in a speech encoder using an order character of LSP (Line Spectrum Pair) parameters in an LSP parameter quantizer using SVQ (Split Vector Quantization) used in a low-speed transmission speech encoder, includes the steps of rearranging a codebook according to an element value of a reference row for determining a range of code vectors to be searched; and determining a search range by using an order character between a given target vector and an arranged code vector to obtain an optimal code vector. The method gives effects of reducing computational complexity required to search the codebook without signal distortion in quantizing the LSP parameters of the speech encoder using SVQ, and reducing computational complexity without loss of tone quality in G.729 fixed codebook search by performing candidate selection and search on the basis of the correlation value size of the pulse position index.

Abstract: An apparatus to process an audio signal includes an encoder to determine a variable window size of sub bands of a frame of an audio signal, to transform the sub bands according to the variable window size from a first domain to a second domain, to quantize the transformed sub bands, and to multiplex the quantized sub bands and information on the variable window size corresponding the respective sub bands.

Abstract: A method and apparatus to extract an important frequency component of an audio signal and a method and apparatus to encode and/or decode an audio signal by using the same. The method of extracting an important frequency component of an audio signal includes converting an audio signal of a time domain into an audio signal of a frequency domain, selecting a frequency band having a harmonic feature from the converted audio signal of the frequency domain, and extracting an important frequency component from the selected frequency band having the harmonic feature.

Abstract: Provided are a frame error concealment method and apparatus and an error concealment scheme construction method and apparatus. The frame error concealment method includes generating a new signal by synthesizing a plurality of previous signals that are similar to a signal of an error frame and reconstructing the signal of the error frame using the generated signal.

Abstract: An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.

Abstract: Disclosed is an apparatus and method for inspection of an electronic device, which can increase the efficiency of inspection by performing a self inspection. According to an aspect of an exemplary embodiment of the present invention, an electronic device is provided with circuit units, comprising an interface unit for interfacing with an external storing apparatus; a user setting unit for performing a self inspection; and a controller for controlling each circuit unit to receive a preset control signal, and for controlling the interface unit to store a responding signal corresponding to the control signal outputted from the respective circuit unit in the external storing apparatus if a first function to perform the self inspection is selected through the user setting unit.

Abstract: A power amplifier based on a voltage-coupling scheme is disclosed. The power amplifier includes two active elements connected to each other via a primary part of a transformer, in which one of the two active elements is connected to a power-supply voltage of the amplifier, the other one is connected to a ground terminal of the amplifier, and an output power appears in a secondary part of the transformer. Therefore, a potential difference between ports of the active elements is decreased, resulting in increased reliability of the power amplifier.

Abstract: An adaptive encoding method includes splitting an input signal into a low-frequency band signal and a high-frequency band signal; performing forward adaptive linear prediction on the low-frequency band signal and thus filtering the low-frequency band signal; selectively performing backward adaptive linear prediction or long-term prediction on the filtered low-frequency band signal according to the analysis result of the low-frequency band signal; transforming the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, into a signal in a frequency domain and quantizing the signal; and encoding the high-frequency band signal using the low-frequency band signal, on which backward adaptive linear prediction or long-term prediction has been performed, or the quantized signal. Therefore, compression efficiency of both speech and music signals can be enhanced, and a robust compression method can be provided for various audio contents at a low bit rate.

Abstract: Encoding and/or decoding a wideband signal produces high frequency band spectra from low frequency band spectral information. Linear prediction filter coefficients are determined for the entire wideband spectrum of an input signal. An energy value in each of a plurality of sub-bands in the high frequency band is determined and encoded. The short-term correlation removed input signal is then down-sampled to form a low frequency band signal. At a decoder, the high frequency band signal is generated using the encoded low frequency band signal. The energy in each sub-band of the high frequency band is adjusted using the encoded energy value. Thus, the spectral envelope for the entire wideband signal is synthesized and decoded using linear predictive synthesis.

Abstract: Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.

Abstract: A method and apparatus to perform bandwidth extension encoding and decoding encodes and/or decodes a high frequency signal using an excitation signal for a low frequency signal encoded in a time domain or a frequency domain or using an excitation spectrum for the low frequency signal. Accordingly, although an audio signal is encoded or decoded using a small number of bits, the quality of sound corresponding to a signal in a high frequency band does not degrade. Therefore, a coding efficiency of the audio signal can be maximized.

Abstract: A method and apparatus to decode audio data constructed with a plurality of layers. An error concealment method of process a decoded bitstream selects one of a frequency domain and a time domain in order to conceal the errors, detects a position where the errors exist in a frame when the error concealment method in the frequency domain is selected, and conceals the errors only in a segment after the detected position.

Abstract: A method of multi-path trellis coded quantization (TCQ) usable in a speech coding system, and a quantizer using the method. Specifically the method includes calculating accumulated distortions corresponding to 2N survivor paths, wherein N indicates an integer greater than two, each of the 2N survivor paths is going towards one of nodes at an i th stage of a trellis, and i indicates an integer greater than zero, comparing the accumulated distortions respectively corresponding to the 2N survivor paths to select N paths among the 2N survivor paths, wherein the accumulated distortions corresponding to selected N paths are smaller than the accumulated distortions corresponding to unselected N paths establishing the selected N paths as survivor paths going toward an i+1 th stage, and selecting an optimal path among the 2N survivor paths corresponding to each node of a last stage.

Abstract: An adaptive time/frequency-based encoding mode determination apparatus including a time domain feature extraction unit to generate a time domain feature by analysis of a time domain signal of an input audio signal, a frequency domain feature extraction unit to generate a frequency domain feature corresponding to each frequency band generated by division of a frequency domain corresponding to a frame of the input audio signal into a plurality of frequency domains, by analysis of a frequency domain signal of the input audio signal, and a mode determination unit to determine any one of a time-based encoding mode and a frequency-based encoding mode, with respect to the each frequency band, by use of the time domain feature and the frequency domain feature.

Abstract: A system for generating and sending a Logical Node identification signal as part of a data stream is disclosed. The system also includes subscriber stations capable of receiving and extracting Logical Node identification information from a data stream. The subscriber stations create new messages including the Logical Node identification signal and send the message to the transmission network system control, such that switching of data streams to an appropriate channel over the information transmission network is done consistent with the network topology.

Abstract: A method of displaying a list of menus on a screen of a mobile telephone is described. Each menu is represented by a menu icon including at least one of a corresponding text and graphical illustration identifying each menu. The method includes the steps of entirely displaying an initial menu icon representing one of the list of menus on the screen, displaying a portion of the initial menu icon and a second portion of an adjacent menu icon on the screen when a telephone user presses down a navigation key or an arrow key in a direction toward the adjacent menu icon, and entirely displaying the adjacent menu icon on the screen. By using the method according to the present invention, an improved user interface of a top or sub menu list display is achieved by displaying the menu list using a scroll technique and by including a numerical and graphical menu indicator indicating the current menu location in a menu tree.

Abstract: A method and apparatus for streaming content to an access network in an interactive information distribution system. The method encapsulates the content in accordance to an Internet Protocol (IP). The content is then transcoded into a format supported by the access network, and streamed over a distribution network to a remote server or to a subscriber terminal that is coupled to the access network. The apparatus is embodied as stream caching server for streaming the content encapsulated within the packet to access networks via a stream distribution network in response to a request for content. A packet processor is coupled to the stream server for processing the encapsulated content within the IP packets into a format native to the access network.