Abstract: A virtual speaker apparatus and a virtual speaker processing method is disclosed. The virtual speaker apparatus uses the closest surround speaker when a virtual speaker to be virtually generated is a back left virtual speaker or a back right virtual speaker.

Abstract: Disclosed is a handheld salinity analyzer including a body, in which a circuit board and a battery are installed, a cell block provided with one end inserted into one end of the body and the other end on which a projecting part provided with a salinity sensor hole and a depressed part provided with a temperature sensor hole are formed, a salinity sensor inserted into the salinity sensor hole and provided with one end connected to the circuit board, a temperature sensor inserted into the temperature sensor hole and provided with one end connected to the circuit board, an ON/OFF switch installed on the end of the body, a light emitting lamp connected to the circuit board, and a body cover connected to the end of the body provided with the ON/OFF switch or to the cell block, and provided with an operating piece to operate the ON/OFF switch.

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: The present invention concerns a DC arc plasmatron which is a plasma generator at the atmospheric pressure having a structure suitable for improving chemical activity of plasma. The DC arc plasmatron according to the invention comprises a rod cathode 110 connected to a DC power supply 1000, and formed to be positioned close to a nozzle to be a negative; a nozzle anode 120 formed on the nozzle to be a positive, being an opening to inject plasma into a functional gap 150 and having an anode inlet 122, wherein the plasma is created by gas discharge for forming the plasma between the cathode and the anode; a functional anode gap formed on the nozzle to be a positive so that either a function gas or a gas mixture is supplied on some part of the passage of the anode to make it react with the plasma forming gas through the nozzle inlet; and a gas supply unit for supplying a plasma forming gas flowing through the nozzle anode from the side of the rod cathode.

Abstract: A method and apparatus of generating a sound field effect is provided. The sound field effect generating apparatus may generate a frequency coefficient that is frequency-transformed from a direct signal, may generate a reflection signal from the frequency coefficient, may generate an output signal using the frequency coefficient and the reflection signal, and may perform an inverse-frequency transform of the output signal.

Abstract: A surround sound virtualization apparatus and method. The surround sound virtualization apparatus may include an audio decoder to perform head-related transfer function (HRTF) filtering, and a time delay unit to provide a time delay to a plurality of output signals of the audio decoder.

Abstract: A display device and methods thereof. A display device may include a current limit control signal generator to output a current limit control signal according to an input display data, a RGB common gamma adjuster to adjust a value of a RGB common gamma setting signal input via a current limit control signal, a gamma divider to divide a RGB common gamma setting signal having a value adjusted by a RGB common gamma adjuster, and/or a gray amplifier to output a gray signal to drive a display screen by amplifying a signal divided by a gamma divider. A display device may have an automatic current limit function applicable to substantially all kinds of display screen, substantially without additional circuits for data calculation.

Abstract: An audio processing apparatus and method for a mobile device are provided. The audio processing apparatus and method may appropriately determine sound source localizations corresponding to a voice signal and an audio signal, and thereby may simultaneously provide a voice call service and a multimedia service. Also, the audio processing apparatus and method may guarantee quality of the voice call service even when simultaneously providing the voice call service and the multimedia service.

Abstract: A coding apparatus including a base layer, a speech quality enhancement layer, and a multiplexer. The base layer filters an input speech signal using linear prediction coding and generates an excitation signal corresponding to the filtered speech signal through a fixed codebook search and an adaptive codebook search. The speech quality enhancement layer searches a fixed codebook using parameters obtained through the fixed codebook search in the base layer, or searches the fixed codebook using a target signal, which is obtained by removing a contribution of a fixed codebook of the base layer and a signal which is obtained by synthesizing and filtering a previous fixed codebook of the speech quality enhancement layer from a target signal for the fixed codebook search of the base layer. The multiplexer multiplexes signals generated by the base layer and the at least one speech quality enhancement layer.

Abstract: A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

Abstract: A block-constrained Trellis coded quantization (TCQ) method and a method and apparatus for quantizing line spectral frequency (LSF) parameters employing the same in a speech coding system wherein the LSF coefficient quantizing method includes: removing the direct current (DC) component in an input LSF coefficient vector; generating a first prediction error vector by performing inter-frame and intra-frame prediction for the LSF coefficient vector, in which the DC component is removed, quantizing the first prediction error vector by using the BC-TCQ algorithm, and by performing intra-frame and inter-frame prediction compensation, generating a quantized first LSF coefficient vector; generating a second prediction error vector by performing intra-frame prediction for the LSF coefficient vector, in which the DC component is removed, quantizing the second prediction error vector by using the BC-TCQ algorithm, and then, by performing intra-frame prediction compensation, generating a quantized second LSF coefficient

Abstract: A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet.

Abstract: Provided are a broadcast receiving apparatus and a firmware upgrade method thereof The firmware upgrade method includes receiving firmware which is transmitted from headend equipment through an Ethernet and upgrading the firmware in the broadcast receiving apparatus. The determination whether to upgrade is based on product information transmitted from the headend equipment.

Abstract: A cable receiver is disclosed that comprises a multimedia channel unit to receive a multimedia signal, and a multimedia processor to process the multimedia signal. The cable receiver further comprises a cable modem to perform data communication with an information provider that provides appendix information about the multimedia signal and a Web server that hosts a predetermined Web page through a cable network. An information processor adds appendix information to the multimedia signal, and a Web browser transmits request information received for the Web page according to a user's selection and processes the received Web page for the user. The cable receiver further comprises a controller to control data transmission of the cable modem, the information processor, and the Web browser to transmit the appendix information and the Web page to the information processor and the Web browser.

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: A memory management method is provided. In the method, a spatial parameter included in an encoding result is represented as a vector of a time slot and a frequency band in a first domain, a temporary matrix is calculated in the first domain by using the difference between vectors of a current time slot and a previous time slot at the same frequency band and then is stored in a memory, and then a matrix needed to decode the encoding result is represented as a matrix for a time slot and a frequency band in a second domain by using the temporary matrix, thereby reducing the load on the memory for storing matrices on which a decoding operation is performed.

Abstract: A signal processing apparatus includes: a communication unit which communicates with another signal processing apparatus storing an encoded first certificate; and a controller which accesses the other signal processing apparatus through the communication unit, copies the first certificate stored in the other signal processing apparatus and decodes the copied first certificate.

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: A method and apparatus for implementing fixed codebooks as a common module are provided. In the method of implementing fixed codebooks of a plurality of speech codecs as a common module, it is possible to include only a part excluding fixed codebooks in a communication terminal or communication system, support various speech codecs without using a chip with high price and high performance, and reduce a memory space that is occupied by the speech codecs by generating a track of a fixed codebook corresponding to a speech codec based on information on the speech codec among the plurality of speech codecs and selecting a codebook vector corresponding to a target signal among codebook vectors constructed with combinations of pulses represented by the generated track. In addition, it is possible to reduce processing complexity as compared with a case of embodying the common fixed codebook module in software by embodying the common fixed codebook module in hardware.

Abstract: Provided is a method of encoding an audio/speech signal, the method including determining a variable length of a frame, that is, a processing unit of an input signal in accordance with a position of an attack in the input signal; transforming each frame of the input signal to a frequency domain and dividing the frame into a plurality of sub frequency bands; and, if a signal of a sub frequency band is determined to be encoded in the frequency domain, encoding the signal of the sub frequency band in the frequency domain, and if the signal of the sub frequency band is determined to be encoded in a time domain, inverse transforming the signal of the sub frequency band to the time domain and encoding the inverse transformed signal in the time domain. According to the present invention, the audio/speech signal may be efficiently encoded by controlling time resolution and frequency resolution.