摘要:
An Internet Set-Top Box (ISTB) and a method of providing wideband IP telephony services using the ISTB are provided, which are capable of implementing wideband voice communication services using a wideband voice codec, controlling a Real-Time Transport Protocol (RTP) packet payload to process a variety of wideband speech frame payloads, and providing high-quality wideband IP telephony services by controlling a jitter buffer to maintain conference call synchronization.
摘要:
A fixed mobile convergence terminal using a wideband voice codec is provided. The fixed mobile convergence terminal includes a communication unit configured to connect to a network, and a control unit configured to download a wideband voice codec identical to a wideband voice codec of an opposite party terminal from a call control server in a call setting with the opposite party terminal through the communication unit, so that a high-quality voice call function is achieved.
摘要:
A packet route management device, a voice over Internet protocol (VoIP) system, and a method of controlling voice call quality. The packet route management device may manage a packet route in an effort to control VoIP voice call based on bandwidth information of packet route devices. Accordingly, the packet route devices may be allowed to process VoIP packets in real time, and thus voice quality can be maintained at a constant level without packet delay or packet loss.
摘要:
A signal processing technology achieved in a signal processing module, which is physically separate from a control module for controlling overall operations of a signal processing apparatus, is provided. Input of new data to a system memory is recognized. Upon the recognition of the input of the new data, the new data is read from the system memory. The new data read from the system memory is written to a local memory. Data sharing between software and hardware is effectively achieved in a system for performing a wideband codec processing using a dedicated hardware.
摘要:
An apparatus, electronic apparatus and method for adjusting jitter buffer is provided. A previous jitter buffer size based on a jitter buffer size determined according to an adaptive jitter buffer size calculation algorithm is applied in predicting a jitter buffer size of future time such that the predicted jitter buffer size is applied to obtain a jitter buffer size of a valid time. The audio quality of the speech transmitted over a packet switched network is enhanced.
摘要:
An apparatus and method for testing conformance of service choreography are provided. The apparatus for testing conformance of service choreography analyzes an architecture and an operation between web services cooperating on a distributed network to test conformance of a choreography application into which the web services are combined, on the basis of a service choreography specification.
摘要:
Provided are a method and an apparatus for decoding an audio signal. A method for decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses includes decoding the encoded audio signal, setting a smoothing frequency band of the decoded audio signal according to a layer structure of the layered sinusoidal pulse coding scheme, dividing the smoothing frequency band into one or more subbands, and smoothing the decoded audio signal on a subband-by-subband basis. Accordingly, a decoding operation time can be reduced and the quality of a synthesized signal can be improved by variably setting a frequency band to be smoothed, when decoding an audio signal encoded by a layered sinusoidal pulse coding scheme using one or more sinusoidal pulses.
摘要:
An encoding apparatus is provided. The encoding apparatus includes a track structure determiner determining a track structure using frequency coefficients, a frequency coefficient allocator allocating the frequency coefficients to each track according to the determined track structure, and a quantizer quantizing one or more pulses in each track based on a number of frequency coefficients allocated to a corresponding track. The encoding apparatus can prevent the degradation of sound quality by avoiding the problem faced by most sinusoidal quantization techniques using a fixed track structure, i.e., a failure to quantize all pulses due to mismatches between the pulse distribution of frequency coefficients and a track structure.
摘要:
A service goal interpreting apparatus for goal-driven semantic service discovery is provided. The service goal interpreting apparatus includes a goal interpretation unit that interprets a goal of at least one service or application provided on the Web, and a goal registration unit that registers the goal interpreted by the goal interpretation unit in a service registry.
摘要:
Disclosed is a coding apparatus and method using residual bits. Accordingly, performance (voice quality) is enhanced by quantizing a full-band gain of frequency coefficients existing in sub-bands to which bits are not assigned in an algebraic vector quantization (AVQ). Further, the performance (voice quality) is enhanced by sequentially quantizing a sub-band gain of sub-bands to which bits are not assigned until residual bits are removed. Furthermore, the performance (voice quality) is enhanced by demodulating AVQ coefficients, and correcting quantization noises starting with a coefficient having the greatest absolute coefficient among the AVQ coefficients, when residual bits additionally remain.