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:
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:
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:
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.
摘要:
An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.
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 and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.
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:
An apparatus and method for testing web service interoperability. The apparatus includes a state model generator generating a state model of a composition system expressing an operation of exchanging a message when web services interoperate with each other as a state machine on the basis of content of a simple object access protocol (SOAP) service description language (SSDL) web service specification, a test structure generator generating a web service test structure by including information about an operating environment in which a web service test is carried out in a composition system structure that includes structured information about web services performing a message exchange operation, a test case generator generating a test case including definitions of execution functions for testing whether or not the web services interoperate with each other according to a message exchange operation sequence defined in the SSDL specification, from the state model of the composition system and information about the web service test structure, and a test case executor inputting a value to the test case to cause test target web services to execute the test case and testing whether the web services interoperate with each other according to the message exchange operation sequence defined in the SSDL specification.