Abstract:
A method for distributing components of a distributed computing system, according to an embodiment of the present invention, includes determining an appropriate number of available server instances, classifying a plurality of components loaded on the distributed computing system into clusters of which the number is equal to the number of the available server instances with reference to interdependent relations among the components, calculating an amount of computing resources required for each of the component clusters classified, rearranging the component clusters to adjust each of the amount of computing resources required for the component clusters to a value within an appropriate range, and deploying the component clusters, of which the computing resource request amounts are adjusted, to the available server instances.
Abstract:
A codec platform apparatus which can perform encoding or decoding regardless of a sampling frequency supported by a codec platform is provided. The codec platform apparatus includes an analog-to-digital converter (ADC) converting an analog input signal into a digital signal by sampling the analog input signal at a codec platform sampling frequency; a sampling frequency converter converting the digital signal provided by the ADC into a digital signal having a codec sampling frequency; and an encoder generating a bit stream by compressing the digital signal provided by the sampling frequency converter. Since there is no need to adopt a new codec platform even when an existing codec platform does not support the sampling frequency of a new codec, there is no need to implant the new codec. Therefore, it is possible to improve user satisfaction.
Abstract:
An encoding apparatus and a decoding apparatus for reducing the quantization error of a G.711 codec and improving sound quality are provided. The encoding apparatus includes a G.711 encoder which generates a G.711 bitstream by encoding an input audio signal; an enhancement-layer encoder which chooses one of a static bit allocation method and a dynamic bit allocation method that can produce less quantization error based on the input audio signal and the G.711 bitstream, and outputs an enhancement-layer bitstream including encoded additional mantissa information obtained by using the chosen bit allocation method; and a multiplexer which multiplexes the G.711 bitstream and the enhancement-layer bitstream. Therefore, it is possible to reduce the quantization error of a G.711 codec and improve sound quality.
Abstract:
Disclosed are a system and method for sharing and retrieving information through a Wiki technology. A Wiki-based information provision system according to the present invention includes a Wiki information input unit for receiving, from a user, a command for writing or changing Wiki information defining a web page written and shared by users included in a group consisting of a plurality of users; and a Wiki server for managing the Wiki information input by the user by using information about associative relationship between the users according to classes or interests of the users. The system according to the present invention may filter a retrieval result using associative relationship between users and first provide information written by a user of the same group as the retrieval result, thus establishing an efficient knowledge collaboration structure.
Abstract:
An apparatus and method for adaptive sub-band allocation of spectral coefficients are disclosed. The sizes of sub-bands are determined according to the distribution of spectral coefficients transformed from an input speech/audio signal to perform more elaborate quantization in units of sub-bands. Thus, quantization noise of the spectral coefficients is reduced, and sound quality in a frequency region is enhanced, thereby improving the quality of the signal.
Abstract:
A wideband Voice over Internet Protocol (VoIP) terminal is provided. The wideband VoIP terminal includes a synchronous serial interface which processes audio data input thereto or output therefrom in series in synchronization with a clock; and an audio accelerator which encodes or decodes the audio data, wherein the synchronous serial interface includes a buffer buffering the audio data and a buffer controller controlling the buffer and the audio accelerator includes a memory storing the audio data processed by the synchronous serial interface under the control of the buffer controller, a memory controller controlling the memory and an encoder/decoder encoding/decoding the audio data. The wideband VoIP terminal can facilitate the input and output of data.
Abstract:
The transmission delay of a voice frame can be reduced by performing internal collision resolution and frame aggregation according to the presence or absence of a voice frame awaiting transmission in a MAC layer, thereby reducing an end-to-end voice transmission delay time for a VoIP service.
Abstract:
A fixed mobile convergence (FMC) communication apparatus using a wideband audio codec is provided. The FMC communication apparatus includes an application processor which is capable of processing one or more wireless communication protocols and supports a wideband audio codec; a wideband audio signal input/output (I/O) unit which is connected to the application processor and processes the input and output of wideband audio signals; a mobile network access unit which is connected to the application processor and accesses a mobile communication network; and a wireless local area network (LAN) access unit which is connected to the application processor and wirelessly accesses an access point (AP). The FMC communication apparatus can access the base station of a mobile internet system or the base station of a mobile communication system. In addition, the FMC communication apparatus can access a Bluetooth device or an AP using a short-range wireless communication method. Moreover, the FMC communication apparatus can provide high-quality voice call services and various multimedia internet services by using a wideband audio codec.